| 00:02.30 | [TK]D-Fender | navaismo: INVITE sip:728996@voiprotel.net SIP/2.0 Contact: <sip:anonymous@10.0.1.103:5060> |
| 00:02.42 | [TK]D-Fender | navaismo: You should follow your own advice about "Asterisk SIP Settings" |
| 00:03.10 | navaismo | contact |
| 00:03.16 | raspberrypifan | ? |
| 00:04.01 | [TK]D-Fender | Clearly improper NAT setup |
| 00:04.30 | navaismo | nein is configured properly |
| 00:05.07 | navaismo | eternhost localnet and everything |
| 00:05.22 | [TK]D-Fender | it's passing the PRIVATE IP and not the public. |
| 00:05.40 | [TK]D-Fender | So whatever you did did not get the job done |
| 00:06.06 | navaismo | nope |
| 00:06.11 | navaismo | is working fine |
| 00:06.18 | [TK]D-Fender | It's still wrong |
| 00:06.27 | navaismo | so my pi is crazy |
| 00:06.32 | [TK]D-Fender | Regardless of if you're lucky enough that they might ignore your Contact: |
| 00:07.06 | raspberrypifan | pi? |
| 00:07.28 | navaismo | yep |
| 00:07.42 | raspberrypifan | ur on a pi |
| 00:09.33 | navaismo | <PROTECTED> |
| 00:09.33 | navaismo | <PROTECTED> |
| 00:09.33 | navaismo | <PROTECTED> |
| 00:09.33 | navaismo | <PROTECTED> |
| 00:09.35 | navaismo | yes |
| 00:09.51 | raspberrypifan | ? |
| 00:11.20 | raspberrypifan | http://asteriskecuador.org/ |
| 00:11.43 | raspberrypifan | hasnt been updated in4 years |
| 00:11.44 | raspberrypifan | sad |
| 00:12.50 | navaismo | yes I use a RPi for that |
| 00:13.08 | raspberrypifan | ive been thining of using my pi for * |
| 00:13.19 | *** join/#asterisk Corey84 (~Corey84@unaffiliated/corey84) |
| 00:14.05 | [TK]D-Fender | [20:09]navaismoExternhost: pbxdm.ath.cx <- go ping it from CLI |
| 00:14.16 | navaismo | going... |
| 00:14.34 | navaismo | amm from my job or from the pi? |
| 00:14.58 | [TK]D-Fender | the server itself |
| 00:15.03 | navaismo | [TK]D-Fender, from the job this is working i have audio so far http://digital-merge.com/ml5/mein.htm |
| 00:15.05 | navaismo | ok going |
| 00:15.44 | navaismo | 10.0.1.103 |
| 00:15.47 | [TK]D-Fender | FAIL |
| 00:16.05 | [TK]D-Fender | That's what you get for horrible hosts planning |
| 00:16.07 | navaismo | from my job-->PING pbxdm.ath.cx (189.241.90.108) 56(84) bytes of data. |
| 00:16.07 | navaismo | 64 bytes from dsl-189-241-90-108-dyn.prod-infinitum.com.mx (189.241.90.108): icmp_seq=1 ttl=248 time=27.2 ms |
| 00:16.24 | navaismo | so my f*** router is messing around |
| 00:16.31 | [TK]D-Fender | Yes, well the server looks at ITSELF to get the IP it gives out |
| 00:16.34 | navaismo | but so far remote extensions are working |
| 00:16.48 | navaismo | even that webrtc link |
| 00:17.09 | [TK]D-Fender | You could be getting lucky NOW... and then use a provider that trusts what you offer and you are dead in the water |
| 00:17.21 | navaismo | "So I Got That Goin' For Me, Which is Nice " |
| 00:17.43 | navaismo | but as I said my domain is going to expire tomorrow no more funny test |
| 00:17.45 | navaismo | :'( |
| 00:18.14 | navaismo | well go to go... 2 hours to travel home have a nice day/night |
| 00:18.19 | navaismo | im going to check that later |
| 00:18.34 | [TK]D-Fender | http://demotivators.despair.com/demotivational/troubledemotivator.jpg |
| 00:29.41 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 01:42.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:08.40 | *** join/#asterisk makubi (~makubi@xdsl-87-78-176-119.netcologne.de) |
| 02:09.05 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 02:39.24 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 02:40.07 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 03:08.28 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-ohvaytjogocrcyqj) |
| 03:09.04 | raspberrypifan | lvlinux |
| 04:06.29 | lvlinux | yeah? |
| 04:06.49 | raspberrypifan | hi |
| 04:06.58 | lvlinux | hey |
| 04:10.08 | raspberrypifan | so remember my project? |
| 04:10.45 | lvlinux | yep |
| 04:11.00 | lvlinux | well shoot i gtg now be back tomorrow... |
| 04:11.15 | raspberrypifan | wait waht time r u coming back |
| 04:11.21 | raspberrypifan | ppl are saying that what u said is impossible |
| 04:12.40 | lvlinux | oh? |
| 04:12.54 | lvlinux | probaly b back around noon eastern, maybe bit before. |
| 04:12.56 | lvlinux | cya! |
| 04:13.04 | lvlinux | it's not impossible btw lol... |
| 04:14.06 | raspberrypifan | with that hardware they said |
| 04:14.08 | raspberrypifan | ok |
| 04:15.46 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 04:35.13 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 04:38.46 | ChkDigit | I just got my new Audiocodes MP-114... Does anyone know how to tell it to stop trying to authenticate with caller-id as the username? |
| 04:38.58 | ChkDigit | ... on incoming calls. |
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| 05:09.05 | *** join/#asterisk bhavikpatel6842 (~root@122.170.90.117) |
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| 06:19.41 | bhavikpatel6842 | Hi All ] |
| 06:19.58 | bhavikpatel6842 | I have an issue with get variable PHP AGI function. |
| 06:20.51 | bhavikpatel6842 | I am using phpagi.php,v 2.14 version |
| 06:22.03 | bhavikpatel6842 | Here is my http://pastebin.com/3zpuFBfw |
| 06:22.23 | bhavikpatel6842 | Can any one help me out of this issue ? |
| 06:24.23 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:24.46 | r00f | i am unfamiliar with php, i use perl with asterisk::agi |
| 06:24.53 | ChannelZ | not sure what the "problem" is |
| 06:25.06 | bhavikpatel6842 | Thanks np |
| 06:25.29 | bhavikpatel6842 | Is there any one else who get any issue in my code or suggest me how to solve this one ? |
| 06:26.39 | ChannelZ | Again, what is the actual problem? I'm not seeing what isn't working or what you're expecting to happen that is not |
| 06:27.40 | bhavikpatel6842 | Here you can see in DumpChan I am getting TOKEN=9f1be342df9e20b7268b221a3cd5ca16 but not getting using $tokenname=$this->obj_agi->get_variable('TOKEN'); |
| 06:28.10 | bhavikpatel6842 | and its getting another variable $trunk_name=$this->obj_agi->get_variable('TRUNKNAME'); and its array also listed in pastebin |
| 06:28.31 | bhavikpatel6842 | so variable mismatch and getting right value in wring variable. |
| 06:28.42 | ChannelZ | oh everything looks shifted |
| 06:28.53 | bhavikpatel6842 | yeah that is the issue. |
| 06:29.08 | bhavikpatel6842 | this is wrong.right ? |
| 06:30.37 | ChannelZ | seems so but I don't use AGI wrappers so I dunno if it's a problem with that thing |
| 06:31.33 | bhavikpatel6842 | I am generating call using callfile. |
| 06:31.33 | ChannelZ | Although all of the vars you are fetching are just channel variables you set presumably, in which case the DumpChan bit isn't necessary. I dunno if something is getting stuck in its return buffer system or what |
| 06:32.23 | bhavikpatel6842 | But DumpChan getting proper so why get_variable AGI function not able to get those variable with proper sequence. |
| 06:34.06 | ChannelZ | hang on I'm trying to look at this stupid phpagi.php crap |
| 06:34.41 | bhavikpatel6842 | Thanks |
| 06:35.21 | bhavikpatel6842 | Do you want my Current PHP AGI file ? |
| 06:37.47 | ChannelZ | yah sure |
| 06:39.14 | bhavikpatel6842 | http://pastebin.com/AtcgEA6X |
| 06:39.41 | ChannelZ | My guess is that this is a bug in phpagi not parsing the 200 response back from the exec properly (or is somehow being sent a duplicate..) |
| 06:40.50 | bhavikpatel6842 | So DO you have any proper AGi file ? |
| 06:44.21 | *** join/#asterisk LiuYan (~liuyan@211.162.64.249) |
| 06:45.30 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 06:46.02 | *** part/#asterisk LiuYan (~liuyan@211.162.64.249) |
| 06:48.32 | ChannelZ | print_r the result of the $this->obj_agi->exec('DumpChan',''); I'm curious to see what it's returning |
| 06:49.02 | bhavikpatel6842 | Yeah you check this http://pastebin.com/3zpuFBfw |
| 06:49.24 | bhavikpatel6842 | I already provided result of DumpChan. |
| 06:49.26 | bhavikpatel6842 | Please check |
| 06:49.27 | ChannelZ | no, you're throwing the return of the exec away |
| 06:50.36 | bhavikpatel6842 | so what function I have to use fore replacement of exec ? |
| 06:51.52 | ChannelZ | $this->obj_agi->verbose("dumpchan exec: " . print_r( $this->obj_agi->exec('DumpChan',''), true ) ); I guess |
| 06:51.59 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 06:52.07 | bhavikpatel6842 | Yeah 1 min giving you |
| 06:52.28 | ChannelZ | and what version of asterisk is this |
| 06:54.08 | bhavikpatel6842 | Nope that way getting Invalid or unknown command |
| 06:54.35 | bhavikpatel6842 | Please check this one : http://pastebin.com/Z40JNV4s |
| 06:55.27 | ChannelZ | what the heck is Dump chain? |
| 06:55.58 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:56.27 | ChannelZ | just do this. Comment out your calling of DumpChan at all and see if the rest of it works. |
| 06:56.40 | bhavikpatel6842 | let me do it.1 min please |
| 06:57.07 | ChannelZ | It seems like this version of phpagi anyway gets confused if a command returns more than 4 lines. |
| 06:57.17 | bhavikpatel6842 | Still same result. |
| 06:57.22 | ChannelZ | or 5 rather. |
| 06:58.01 | bhavikpatel6842 | I have to add 5 custom variable while call start and get those varible while answer the call. |
| 06:58.11 | bhavikpatel6842 | this issue I am facing first time. |
| 06:58.41 | ChannelZ | yes but you don't need dumpchan to see them. |
| 06:59.09 | bhavikpatel6842 | I just wanted to show you that dumpchan also showing proper value. |
| 06:59.40 | bhavikpatel6842 | and also set properly in channel although not getting right value in right vars. |
| 07:01.24 | ChannelZ | All I can say is that the return of *something* (my guess is the DumpChan) isn't completely getting read out of the stdin buffer, which is then throwing everything off by 1. So when you get_variable, it's actually reading the response of the previous command. |
| 07:01.55 | ChannelZ | turn on AGI debug and you should see the entire command conversation |
| 07:02.14 | bhavikpatel6842 | okey giving you agi debug |
| 07:03.16 | bhavikpatel6842 | http://pastebin.com/haXG54sF |
| 07:04.52 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 07:04.58 | ChannelZ | you seem to be sending a blank command which is where the 510 is coming from |
| 07:06.04 | ChannelZ | but anyway you can see it's sending the correct responses, but phpagi is not parsing the return correctly and is off by one response. |
| 07:06.43 | bhavikpatel6842 | Yeah |
| 07:07.03 | ChannelZ | can you post the *actual* script you are using right now (demo.php) |
| 07:07.14 | ChannelZ | or sorry ivr.php |
| 07:07.15 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 07:08.01 | bhavikpatel6842 | first my ivr.php call and then call another file class function and get directly vars. |
| 07:08.33 | ChannelZ | oh I see.. so yes, it is demo.php |
| 07:08.46 | ChannelZ | oh wait |
| 07:08.54 | bhavikpatel6842 | Yeah both file have same value just name is idff |
| 07:08.56 | bhavikpatel6842 | diff* |
| 07:09.13 | ChannelZ | so is ivr.php doing an "exec agi(demo.php)" or similar? |
| 07:10.17 | bhavikpatel6842 | No my dialplan call Ivr.php and then |
| 07:10.17 | bhavikpatel6842 | include "include/class/class_main_ivr.php"; |
| 07:10.17 | bhavikpatel6842 | $agi = new AGI(); |
| 07:10.17 | bhavikpatel6842 | $ivr_obj = new main_ivr($agi); |
| 07:10.40 | bhavikpatel6842 | and in class_main_ivr file contains code for get_variable function. |
| 07:11.34 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:12.10 | ChannelZ | I'm confused where <SIP/6060-0000000f>AGI Tx >> agi_request:demo/demo.php is coming from then |
| 07:12.32 | bhavikpatel6842 | I have two folder |
| 07:12.54 | bhavikpatel6842 | demo/demo.php |
| 07:12.55 | bhavikpatel6842 | second one webservice_ivr/ivr.php |
| 07:13.06 | bhavikpatel6842 | both version have same code so no issue in that. |
| 07:13.28 | bhavikpatel6842 | just folder and file name diff apart from other same thing in both source. |
| 07:13.57 | ChannelZ | I'd have to see your complete code to know what's going on for sure but it seems like you're either trying to run two AGIs and that's where the responses are getting thrown off, or phpagi is broken in some other fashion causing the same result, the responses are not being read correctly. |
| 07:14.28 | bhavikpatel6842 | Okey sending you. |
| 07:14.37 | ChannelZ | asterisk IS returning the right data and responses to the commands it's being sent, it's your phpagi wrapper that is broken |
| 07:15.06 | *** join/#asterisk RadJackson (~RadJackso@cpe-xd001622.cust.jaguar-network.net) |
| 07:15.34 | bhavikpatel6842 | Yeah I know asterisk showing also right data but phpagi sending wrong value |
| 07:17.08 | RadJackson | Hi , i'm unable to make outgoing call with an anonymous CallerID, my provider asked me to add header Privacy:id , p-preferred-identity , and finally CALLERPRES()=prohib. Here is what i did , it is still not working : http://pastebin.com/GFVh50W7 , any idea ? |
| 07:20.18 | bhavikpatel6842 | @ChannelZ Please check this http://pastebin.com/S6S0zG9h |
| 07:20.20 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:7073:60d2:d62e:f30b) |
| 07:20.27 | ChannelZ | well I'd assumne you still need to set P-Asserted-Identity to an actual value but I have no idea |
| 07:20.33 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 07:25.21 | bhavikpatel6842 | Any changes in my code ? |
| 07:26.54 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 07:27.16 | ChannelZ | I don't see anything obvious.. this is the exact script you ran when you pastebin'd the AGI debug? Because I don't know where the null command is coming from |
| 07:27.35 | ChannelZ | but that's where it's getting thrown off I think. |
| 07:27.41 | bhavikpatel6842 | Yeah I have same code which I provided you. |
| 07:27.41 | ChannelZ | What version of asterisk is this? |
| 07:27.45 | bhavikpatel6842 | no changes in script |
| 07:27.56 | bhavikpatel6842 | Asterisk 1.8.26.1 |
| 07:29.02 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 07:29.55 | bhavikpatel6842 | Do you think its an asterisk issue ? |
| 07:30.06 | bhavikpatel6842 | Because I tested 2 system and same issue. |
| 07:30.16 | bhavikpatel6842 | so may phpagi or some thing else issue. |
| 07:31.09 | ChannelZ | I don't really know. After the AGI starts and Asterisk sends all the AGI variables, in your output I see |
| 07:31.12 | ChannelZ | <SIP/6060-0000000f>AGI Rx << |
| 07:31.39 | bhavikpatel6842 | Its strange issue. |
| 07:31.41 | ChannelZ | and then it returns a 510 invalid command. But I don't see where this blank command is being sent |
| 07:32.00 | ChannelZ | unless there's a hidden character in your first get_variable command, a carrige return or something |
| 07:32.05 | bhavikpatel6842 | <SIP/6060-0000000f>AGI Tx >> |
| 07:32.05 | bhavikpatel6842 | <SIP/6060-0000000f>AGI Rx << |
| 07:32.05 | bhavikpatel6842 | <SIP/6060-0000000f>AGI Tx >> 510 Invalid or unknown command |
| 07:32.29 | bhavikpatel6842 | Yes why its returing 2 tx and rx blank value. |
| 07:32.31 | skrusty | it is an invalid command |
| 07:32.35 | skrusty | there's nothing there |
| 07:32.52 | skrusty | extra newline? |
| 07:32.57 | ChannelZ | The first Tx is the blank line signifying the end of the agi variables. This is from the point of view of Asterisk sending to your script |
| 07:32.58 | bhavikpatel6842 | but not getting what is "Rx" and "tx" |
| 07:33.11 | ChannelZ | The Rx is a command it has received from your script. |
| 07:33.15 | skrusty | receive and transmit |
| 07:33.24 | ChannelZ | yes |
| 07:33.52 | bhavikpatel6842 | So not getting where this 510 Invalid or unknown command getting in my script. |
| 07:34.05 | bhavikpatel6842 | Do you want to show my call file ? |
| 07:34.07 | ChannelZ | that's the problem |
| 07:34.49 | skrusty | bhavikpatel6842: are you correctly handling the end of vars newline? |
| 07:34.50 | ChannelZ | your script isn't handling the response. So when you do the first "real" command, the first response you get is actually the 510 from the command you didn't send. I just don't know where that 'command' is coming from |
| 07:35.03 | ChannelZ | My only guess initially is a hidden character or something |
| 07:35.34 | bhavikpatel6842 | Here is my call file : http://pastebin.com/imSWG5tA |
| 07:35.49 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 07:36.29 | ChannelZ | that doesn't matter. This OH |
| 07:37.55 | ChannelZ | make sure you don't have any blank lines before your <?php in your script |
| 07:38.11 | bhavikpatel6842 | Let me check |
| 07:38.40 | ChannelZ | it could be a crlf even. Is your script saved with "unix newlines"? \n only |
| 07:38.56 | ChannelZ | depending what editor you're using |
| 07:39.04 | ChannelZ | But I bet that's where the blank command is coming from |
| 07:39.19 | bhavikpatel6842 | yeah |
| 07:40.20 | bhavikpatel6842 | I removed all blanks line still same result. |
| 07:40.44 | ChannelZ | What editor are you using? |
| 07:40.54 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 07:41.01 | bhavikpatel6842 | kwrite |
| 07:41.09 | bhavikpatel6842 | I am using fedora 19 |
| 07:41.15 | skrusty | how are you terminating your commands? |
| 07:41.18 | ChannelZ | well make sure it's not saving CRLF |
| 07:41.24 | *** join/#asterisk DataWraith (~s1gny@178-27-221-137-dynip.superkabel.de) |
| 07:41.38 | bhavikpatel6842 | Not much idea about CRLF. |
| 07:41.45 | bhavikpatel6842 | Can you please tell me what is that ? |
| 07:41.47 | ChannelZ | skrusty: he's using the phpagi wrapper so it's doing \n its self |
| 07:42.03 | skrusty | ok |
| 07:42.42 | ChannelZ | bhavikpatel6842: I don't know anything about kwrite but you can save text files in most editors with "Windows/DOS" type newlines which is CRLF (carrige return, linefeed) or "Unix" which is LF only |
| 07:43.25 | bhavikpatel6842 | Okey 1 min save file in windows and trying. |
| 07:43.33 | ChannelZ | I'm guessing it's CRLF and the CR is being output by PHP to stdout which is the invisible command Asterisk is getting. |
| 07:43.40 | ChannelZ | no, you don't want CRLF. |
| 07:43.46 | skrusty | heh :) |
| 07:43.47 | ChannelZ | You want LF only |
| 07:44.28 | bhavikpatel6842 | Guys what I have to do know ? |
| 07:45.12 | ChannelZ | it might even be a blank line at the head of any of your sub-files, the ones being included() in your main_ivr script |
| 07:45.27 | ChannelZ | or a blank line at the end, after the closing ?> of any of them |
| 07:45.54 | ChannelZ | probably the latter. |
| 07:45.54 | bhavikpatel6842 | Okey let me remove that all files blank lines |
| 07:46.13 | ChannelZ | sorry I forgot about those. |
| 07:46.35 | ChannelZ | hopefully kwrite being in linux isn't writing CRLF. I'm sure it's a blank line in one of the included scripts. |
| 07:48.23 | ChannelZ | Ugh. what a mind-f*. Wish that'd occured to me half an hour ago, I forgot. |
| 07:49.26 | bhavikpatel6842 | I think something getting properly. |
| 07:49.36 | bhavikpatel6842 | let me test one more times. |
| 07:50.31 | bhavikpatel6842 | Thanks guys for your support. |
| 07:50.56 | ChannelZ | sorry I didn't get that earlier. I just wasn't thinking clearly. Past my bedtime.. |
| 07:50.57 | bhavikpatel6842 | its working and issue is I think my config file contains blank like in lower side of the file. |
| 07:51.06 | bhavikpatel6842 | Hehehe ;) |
| 07:51.30 | ChannelZ | Yeah remember with PHP anything outside of <?php ?> is just output to stdout as text. And AGI is an stdin/stdout interface. |
| 07:52.20 | bhavikpatel6842 | Okey thanks ChannelZ |
| 07:52.32 | ChannelZ | sure have fun |
| 07:52.39 | bhavikpatel6842 | One more thing |
| 07:52.48 | bhavikpatel6842 | is this function right ? |
| 07:52.48 | bhavikpatel6842 | $this->obj_agi->get_variable('CHANNEL(dahdi_span)') |
| 07:55.04 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 07:55.19 | ChannelZ | I think someone asked this before. |
| 07:55.55 | bhavikpatel6842 | I just wanted to use and not sure its right or not. |
| 07:56.02 | bhavikpatel6842 | I use this first time |
| 07:56.19 | ChannelZ | I think it does. |
| 07:57.30 | ChannelZ | I think if you want to use expressions you have to use 'get full variable' which that version of phpagi doesn't have, but simple channel functions should work with normal 'get variable' |
| 07:58.12 | *** join/#asterisk r00f (~r00f@94.200.97.245) |
| 08:00.24 | bhavikpatel6842 | yeah I have that function. |
| 08:01.09 | bhavikpatel6842 | And Like if you use callfile then we can set answer context. |
| 08:01.09 | bhavikpatel6842 | let me know if I am wrong. |
| 08:08.48 | *** join/#asterisk Rumbles (~Rumbles@mxgw1-brh.antmarketing.com) |
| 08:09.10 | bhavikpatel6842 | So can generate call multipal and set answer context using AMI (Asterisk manager) |
| 08:11.22 | skrusty | use AMI originate maybe and you can set a channel and then the extension and context that the answered channel will be passed to? |
| 08:11.57 | bhavikpatel6842 | Do you have some example how to set extension and context ? |
| 08:12.21 | bhavikpatel6842 | because as far as I know AMI onlly just originate channel. |
| 08:22.09 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 08:24.13 | Rumbles | Hi, I'm trying to figure out the cause for an issue we are experiencing on a few of our asterisk servers, one in particular appears to see the issue upwards of 10 times a day, the server is running on asterisk 1.2.30 (I know it's an old version - we are forced to use this at present) along with vicidial, the server hosts a number of conferences, and during the issue we see all the voice chat ongoing on the affected server dropping out. I am |
| 08:24.14 | Rumbles | <PROTECTED> |
| 08:29.11 | Rumbles | We have a number of servers that aren't affected by the issue also which use the same vicidial and asterisk setup, and have no issues. I am not sure whether the issue is a hardware one, or a software one (I am leaning towards asterisk being at fault), as such I have tried changing the network interface in use on the server (different port on the same card) and switch it is connected to. I am just stumped as to why the server seems to lose c |
| 08:29.12 | Rumbles | onnection to the hardware phones as well as the main asterisk server that passes the calls through to it while the netowkr connections are still valid and working. |
| 08:29.57 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 08:30.22 | Rumbles | Also, during the time we see the issue, CPU usage does not spike, nor does memory usage or active prcesses (we are not hitting a maximum process limit on the server), the hard drives are not full... I'm at a loss of things to check. Any suggestions welcome. :) |
| 08:36.35 | WIMPy | You don't really expext anything other than "upgrade", do you? |
| 08:36.49 | Rumbles | Sadly no, but that's not possible |
| 08:37.20 | WIMPy | You might be hitting the CPU limit without seeing it. |
| 08:39.11 | Rumbles | load average on the server never gets above 0.2 |
| 08:39.55 | WIMPy | I don't remember if you see it on the load, but you will normally not see the CPU time. |
| 08:41.21 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 08:41.58 | Rumbles | the server will drop the calls when there are only 4 people connected on it, but it can handle up to 30 people, and quite frequently does, the last time it happened was at 8:30 this morning, since then another 10 people have started work and they are working fine on it currently |
| 08:42.11 | WIMPy | Try to compare the number of active conferences and participants. |
| 08:42.21 | Rumbles | I have that graphed :) |
| 08:42.40 | Rumbles | well, I have the number of active conferences graphed |
| 08:42.59 | *** join/#asterisk DataWraith (~s1gny@178-27-221-137-dynip.superkabel.de) |
| 08:43.18 | WIMPy | Well, ok, that must be some network issue then. But noone will be able to tell you wich of the thousands of bugfixes of the last 7 years is the one you are seeing. |
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| 08:44.16 | Rumbles | I monitor the meetme command, and grep for 0003 (number of conferences with 3 active channels) as well as the total number of conferences (same command grepping for Static) and these both drop to 0 when the issue hits |
| 08:44.54 | Rumbles | so it's an asterisk network issue you think WIMPy ? |
| 08:44.56 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 08:45.49 | WIMPy | More like a guess, but it doesn't seem to be meetme, which I thought might be the one. |
| 08:46.31 | Rumbles | so because the number of conferences drops to zero you think it's a network issue? |
| 08:47.30 | WIMPy | That's why I think it's not the conferences. So what's left then? |
| 08:48.04 | Rumbles | :) thanks |
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| 09:36.11 | Rumbles | WIMPy, I just started to look into scratch building a later version of asteris when someone reported that the sound had died again... earlier I connected a phone on my desk to that server to see if I could continue to ring out when the issue hit, I was able to start a call fine... so I thought I would check the meetme status |
| 09:36.31 | Rumbles | initially, when I ran asterisk -rx "meetme" it returned no info |
| 09:36.49 | Rumbles | after around 3-0 seconds though I just saw "disconnected from asterisk server" then this: http://pastebin.com/VYfNLmWj |
| 09:37.33 | Rumbles | I also see this in the asterisk logs: May 7 10:27:07 WARNING[3739] asterisk.c: No more connections allowed |
| 09:40.02 | WIMPy | You must have some dead console connections, I guess. |
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| 09:42.57 | Rumbles | hmmm I guess, but why is it refusing my new connections :/ |
| 09:44.55 | WIMPy | Must be a bug. |
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| 09:51.51 | Rumbles | :) |
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| 10:42.59 | file | there is a fixed maximum number of console connections, and it is 128 |
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| 10:43.23 | Ice_Strike | Hi |
| 10:44.10 | r00f | didnt know that, tbh. but getting 128 connections is not that easy. some cron job may be looping or hanging |
| 10:44.47 | Rumbles | looking at the logs I see Remote UNIX connection over and over again until I see it tell me that No more connections aloud.... |
| 10:45.05 | Rumbles | or even allowed |
| 10:45.39 | Rumbles | so it looks like something is connecting a huge amount of console connections before the server gives up and drops everytihng :/ |
| 10:45.42 | Ice_Strike | Hey WIMPy |
| 10:46.08 | WIMPy | o/~ |
| 10:47.15 | Ice_Strike | WIMPy Had electrician today, remember we talked about humming noise few weeks ago |
| 10:47.30 | WIMPy | yes |
| 10:48.04 | Ice_Strike | they done re-earthing and testing... |
| 10:48.14 | Ice_Strike | Said all fine now but buzzing still exist :/ |
| 10:48.27 | Ice_Strike | They said it will take a week to disspear. |
| 10:48.39 | Ice_Strike | disappear |
| 10:49.32 | WIMPy | Errr?!? |
| 10:49.42 | WIMPy | Does t need to fade out slowly or what? |
| 10:49.52 | WIMPy | it |
| 10:49.52 | r00f | very slow electrons |
| 10:50.01 | r00f | need to wait for them to go away |
| 10:50.45 | Ice_Strike | I have no idea |
| 10:51.54 | WIMPy | Sounds like snake oil. |
| 10:52.47 | Ice_Strike | Really annoying. |
| 10:53.05 | Ice_Strike | I don't know if they are bullshitting. |
| 10:56.37 | Ice_Strike | r00f really? |
| 10:57.33 | r00f | just trying to bring in some jokes. try to forgive me if it is not allowed here |
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| 11:12.17 | Rumbles | sarcasm doesn't communicate very well in text based commnication :) |
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| 11:45.18 | FlashDel | hi! can somebody tell me how i can force asterisk to NOT send sip OPTIONS? |
| 11:47.14 | WIMPy | Don't tell it to do so. |
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| 11:57.52 | michael_work | qualify=no? or am i wrong? |
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| 12:05.25 | FlashDel | my problem ist, that asterisk is sending faulty OPTIONS to my Fritzbox and that causes timeouts after a while. The fritzbox bans the ip adress of the asterisk because it detects a sip crute force attack |
| 12:06.29 | FlashDel | i am using Asterisk 11.9.0, btw |
| 12:07.50 | WIMPy | hasn't had any issues so far. |
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| 12:09.24 | FlashDel | WIMPy which version do you use? |
| 12:10.07 | WIMPy | 11.9.0 |
| 12:10.28 | FlashDel | mhhh |
| 12:11.12 | WIMPy | Still: If you don't want it, don't switch it on. |
| 12:13.08 | FlashDel | here is a trace if you want to take a look http://pastebin.de/124702 |
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| 12:18.43 | FlashDel | WIMPy, if i turn it out, the registration fails, so i have to turn it on. But i get a timeout after a few minutes. If i set qualifyfreq=600 i get the timeout after a few hours |
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| 13:01.21 | Gugge | FlashDel: registration does in no way require qualify |
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| 13:01.50 | Gugge | FlashDel: but without qualify you have to register more often than your udp timeout in any statefull firwalls |
| 13:04.37 | leifmadsen | that. |
| 13:04.54 | leifmadsen | sounds like the options are just keeping your firewall ports open |
| 13:06.21 | [TK]D-Fender | That's the usual goal... |
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| 13:06.55 | leifmadsen | if you're wanting to talk to people anyways |
| 13:08.17 | FlashDel | mhhh guess you`re right. I am running into timeouts, it doesn`t matter what i am configuring :-((( |
| 13:09.12 | FlashDel | sometimes it works for a few minutes and then these timeouts appear again |
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| 13:10.02 | FlashDel | i configured my asterisk server as exposed host in the fritzbox, so firewalling shouldn`t be a problem |
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| 13:24.20 | leifmadsen | make sure you don't have any ALG stuff turned on :) |
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| 13:49.53 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 13:49.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 14:25.21 | igcewieling | Switching to SIP/TCP should help with firewall timeout stuff. |
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| 14:51.01 | puzzled | FlashDel: I had problems with my Fritzbox 7170 and 7360. They grab port 5060 no matter what |
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| 14:54.02 | puzzled | FlashDel: if you have an issue with an Asterisk box behind a Fritzbox and clients timing out then here's a solution: http://blog.laimbock.com/2014/03/27/how-to-make-asterisk-work-behind-fritz-box/ |
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| 15:19.39 | leifmadsen | step 1) replace your fritz*box :) |
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| 15:34.26 | dan_j | Hi. I'm looking at adding some redundancy. I've been looking at different ways to implement it. From what i can see, I can either use Opensips in front of multiple asterisk servers, or I can use DUNDi with a bunch of asterisk servers. |
| 15:34.45 | dan_j | Does anyone have any experience of either? |
| 15:35.26 | dan_j | Or has anyone tried one of the above solutions and found it to be too problematic? |
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| 15:56.11 | navaismo | dan_j, define redundancy, for trunks, for disks, for the servers what? I have used the HA solution based on the digium Rseries, and tested opensips as a load balancer |
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| 15:59.52 | dan_j | redundancy so that an asterisk server can fail and calls still flow. |
| 15:59.53 | dan_j | sip only |
| 16:00.16 | file | define "still flow" |
| 16:00.19 | dan_j | fail = asterisk stops running for whatever reason |
| 16:00.49 | dan_j | still flow = the sip peers don't notice any problem (or at least 2 or 3 minutes downtime maximum) |
| 16:05.06 | navaismo | dan_j, somewhere in the asterisk wiki lives an exmaple of asterisk SCF with HA and the elderly said that the call never drops(MYTH) |
| 16:06.29 | file | it wasn't a myth, we did it |
| 16:06.43 | dan_j | I know calls will drop when asterisk crashes, thats fine. but i want to ensure that the user can simply redial and be connected. |
| 16:07.32 | navaismo | file, exist a video or something that can validate that myth? |
| 16:07.56 | file | may be one on Youtube, we did it at an AstriCon in front of everyone |
| 16:08.17 | dan_j | Do you remember the year? |
| 16:08.23 | file | nope |
| 16:08.32 | file | SCF is no longer worked on regardless |
| 16:08.56 | navaismo | file-->http://forums.crackberry.com/attachments/blackberry-10-os-f269/214226d1382728776t-feature-request-timeshift-motion-gif-3qxl6a.jpg |
| 16:09.19 | dan_j | I like the look of this document. Although i've not got to testing it yet. Just wanted to see if there was a tested/better way. http://www.ntegratedsolutions.com/wp-content/uploads/2012/07/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf |
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| 16:11.20 | dan_j | navaismo: Page not found |
| 16:11.31 | dan_j | ah, ignore that |
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| 16:54.19 | dan_j | So, ignoring scf. Out of the DUNDi and/or OpenSIPS implmentations, which has the least amount of problems? |
| 16:58.51 | navaismo | your setup for sure |
| 16:59.01 | dan_j | eh? |
| 16:59.19 | navaismo | I mean which one you are more comfortable |
| 16:59.29 | navaismo | and know it better |
| 16:59.34 | dan_j | I'm wondering if i even need opensips if i can use DUNDi |
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| 17:00.32 | navaismo | IMO i would rather opensips |
| 17:01.46 | dan_j | I assume that if i'm using queues, the agents and calls need to go through a single asterisk server? |
| 17:03.10 | *** join/#asterisk Meaty (c772ea12@gateway/web/freenode/ip.199.114.234.18) |
| 17:04.15 | pabelanger | well, there is no easy way to do distributed queues ATM. |
| 17:04.24 | pabelanger | Some state information is just not replicated |
| 17:04.34 | pabelanger | like caller position in side the queue |
| 17:10.10 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 17:10.34 | Meaty | Hi ! Im running Asterisk 1.6.0.28-samy-r117 . Im using queue and sip accounts whithout reinvite. When a phone answer a call, the server add a jitter (too much) on all outgoing streams (outgoing of the server) on each legs of the call. If a phone take a call bypassing the queue, the call is 100% clean. Any one know why the queue application can add this jitter ? Thanks |
| 17:11.02 | pabelanger | are you transcoding audio? |
| 17:11.44 | Qwell | ~asterisk versions |
| 17:11.44 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 17:11.59 | Qwell | ~upgrade asterisk |
| 17:11.59 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:12.02 | Qwell | that's the one |
| 17:12.44 | Qwell | ~trixbox |
| 17:12.45 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 17:12.46 | Qwell | also that |
| 17:14.21 | Meaty | Haha nice |
| 17:15.44 | Meaty | My client is using a Trixbox Pro, i can't upgrade and the fonality support dont found the probem. And no, im not transacoding the audio. |
| 17:16.10 | Qwell | Well, you can't get help here without upgrading... |
| 17:16.59 | Meaty | Np, i understand. Thanks |
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| 18:03.13 | lvlinux | raspberrypifan: hey u here? |
| 18:07.43 | jmmills | Anyone have a recommendation on the simplest solution to "spoofing" a call into a pbx to test a dialplan/agi |
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| 18:09.16 | leifmadsen | jmmills: originate command on the console? |
| 18:09.32 | leifmadsen | could use a context with a local channel that loops back in based on some data |
| 18:10.58 | jmmills | Like originate with a Dial(Local/$spoof@context) |
| 18:11.39 | jmmills | that works for dialplan contexts, but perhaps not for agi |
| 18:11.53 | jmmills | I'd have to force a bunch of channel variable data |
| 18:13.51 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-101-Washington.hfc.comcastbusiness.net) |
| 18:16.38 | [TK]D-Fender | jmmills: call-file <- |
| 18:16.56 | jmmills | like via the spool |
| 18:17.08 | jmmills | Yeah, that's right. I can set channel variables with those |
| 18:17.22 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 18:26.42 | [TK]D-Fender | Correct |
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| 18:33.48 | KNERD | how can I perform a "male menuselect" without it going into the menu interface and instead just exit? |
| 18:36.17 | pabelanger | menuselect/menuselect --enable DONT_OPTIMIZE menuselect.makeopts |
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| 18:36.35 | pabelanger | for more examples: http://svnview.digium.com/svn/testsuite/bamboo/trunk/bin/build-asterisk-only.sh?view=markup |
| 18:36.58 | KNERD | i understand that |
| 18:37.30 | KNERD | however, menuselct does not exist until you do a :"make menuselct" |
| 18:37.53 | *** part/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 18:39.38 | KNERD | thus when I do a "make menuselect" I don't want the infeface to show up up |
| 18:40.44 | KNERD | oh...i see... menuselect.makeopts |
| 18:40.53 | KNERD | make menuselct.makeopts |
| 18:40.55 | KNERD | thanks |
| 18:48.20 | leifmadsen | also documented in the book :) |
| 18:53.34 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 18:53.47 | KNERD | ~book |
| 18:53.47 | 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 |
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| 19:13.54 | BCrookAtRA | I'm trying to play a recording to my callers that contains all ten touchtones (the goal is to trick autodialer robots into thinking we pressed whatever digit they say to press to be removed). But the touchtones always come out sounding clipped as if they were too loud, no matter how quiet I make them, or howlong i play them. Does asterisk 'intercept' touchtones somehow? CIT tones play normally as expected even though they are significntly lo |
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| 19:32.16 | [TK]D-Fender | How are you making them to play back? |
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| 20:05.47 | dan_j | pabelanger: Are there still plans to replicate call queue information? Towards the end of this video staring Kevin Fleming, he said that it wasn't far off. That was in 2008. https://www.youtube.com/watch?v=3ug3sWp5ABA |
| 20:10.00 | pabelanger | dan_j, To my knowledge there is no way to replicate that data right now. Unless there is an external project working on it. |
| 20:10.18 | pabelanger | That's the reason we started writing something ourself, to do just that |
| 20:10.47 | dan_j | But its been discontinued, correct? |
| 20:11.05 | pabelanger | what has? |
| 20:11.23 | dan_j | You mentioned SCF? |
| 20:11.35 | [TK]D-Fender | long dead |
| 20:11.38 | [TK]D-Fender | never released |
| 20:11.47 | [TK]D-Fender | purge from memory |
| 20:11.55 | pabelanger | not me, somebody else did. But ya, that is a dead project. But it wasn't related to queues IIRC. They were doing voice |
| 20:12.11 | navaismo | o7 |
| 20:12.54 | dan_j | So when you said "we started writing something ourselves" what were you talking about? |
| 20:13.14 | pabelanger | dan_j, http://www.youtube.com/watch?v=9aRQUYvzzso |
| 20:13.20 | pabelanger | that was my talk last year at astricon |
| 20:14.34 | dan_j | thanks for that. |
| 20:15.08 | pabelanger | dan_j, since the talk we have expanded a bunch of stuff, but right now the easiest way to see a demo is use: https://github.com/kickstandproject-dev/grunt |
| 20:15.35 | pabelanger | plan to push up some ARI code in the coming weeks for some basic FIFO queues |
| 20:15.40 | pabelanger | outside asterisk |
| 20:18.58 | dan_j | looking forward! I'll take a look at that link tomorrow? |
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| 20:35.02 | dan_j | Not sure why I ended that sentence with a ? :-S |
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| 21:03.08 | *** join/#asterisk swiss__ (~swiss__@70.184.40.66) |
| 21:03.15 | swiss__ | hello all |
| 21:03.26 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 21:03.29 | cj | hey guys |
| 21:03.50 | swiss__ | ahoy hoy |
| 21:04.10 | cj | carrar: hey, we've got some customers coming on line. Do you have any time to help me configure up this cisco 2801 with a VIC-4FXS card? |
| 21:04.24 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:04.38 | swiss__ | well beyond my abilities im afriad, im also here to ask a question |
| 21:04.49 | cj | I'm hoping to feed the SIP from threshinc through this thing and give the customer 4 lines |
| 21:06.38 | cj | swiss__: heh, I give carrar a little money every month for service and I'm hoping I can wheedled him in to giving me a bit of support, too ;-) |
| 21:07.59 | swiss__ | haha nice |
| 21:11.19 | swiss__ | so im trying to get an extensions.conf going that will ring a number of cellphones until someone answers and presses 1 to acknowledge. here is my extensions.con: http://pastebin.com/5sT8zVRg and the result when i place a call: http://pastebin.com/k83zhVNu |
| 21:11.32 | swiss__ | i get 1 call but it never calls the follow up numbers |
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| 21:14.56 | ChannelZ-Wk | your priorities in [findme] look a bit messed up |
| 21:17.13 | ChannelZ-Wk | but anyway you need to use the g flag in your dials to make it continue on in the dialplan |
| 21:18.51 | ChannelZ-Wk | assuming the first person is answering the call but refusing to accept it and just hangs up |
| 21:19.00 | swiss__ | i apologize im still very green with the way extensions.conf works. can you give me a g flag example and how the priorities should be done? what i have there is based on my best guesses from reading various wikis |
| 21:19.13 | swiss__ | yes that is how im testing the first call |
| 21:20.20 | ChannelZ-Wk | well you have extension 1,n and then 1,2 and 1,3. Really you want to specify the first priority specifically, 1,1 and then do 1,n for the others (which will automatically do sequential 2, 3, etc.) |
| 21:20.30 | swiss__ | i see |
| 21:21.04 | ChannelZ-Wk | So you have it backwards basically. Specify priority 1, then use n for the others for that extension. |
| 21:21.37 | swiss__ | alright fixed that |
| 21:23.27 | swiss__ | so i tried again and same thing |
| 21:23.46 | ChannelZ-Wk | You might also have to add some logic to check the status of the previous attempt; If for instance the first person answers and accepts the call, they finish a conversation and the remote end hangs up but the caller doesn't, the dialplan will continue on and call the next guy. |
| 21:23.55 | ChannelZ-Wk | Did you add 'g' to the Dial() options? |
| 21:24.10 | swiss__ | exten => 1,n,Dial(SIP/voipms/9999${findme1},25,g) |
| 21:24.12 | swiss__ | like that? |
| 21:24.22 | ChannelZ-Wk | yeah |
| 21:24.51 | swiss__ | and what about on the first line? the _9999XXXXXXXXXX,1,Dial(SIP/voipms/${EXTEN:4},40,rM(screen)) |
| 21:24.54 | swiss__ | ? |
| 21:25.24 | ChannelZ-Wk | Well that one you don't need to continue in this fashion |
| 21:25.51 | swiss__ | makes sense |
| 21:26.07 | ChannelZ-Wk | It's not really related to this |
| 21:26.22 | swiss__ | [findme] |
| 21:26.22 | swiss__ | exten => s,1,NoOP;(I'd rather the callers just hear ringing so they don't know they are calling something else) |
| 21:26.23 | swiss__ | exten => 1,n,Dial(SIP/voipms/9999${findme1},25,g) |
| 21:26.23 | swiss__ | exten => 1,n,Dial(SIP/voipms/9999${findme2},25,g) |
| 21:26.23 | swiss__ | exten => 1,n,Dial(SIP/voipms/9999${findme3},25,g) |
| 21:26.29 | swiss__ | so thats what ive got now |
| 21:26.50 | swiss__ | make sense or do i still have the priorities wrong? |
| 21:27.16 | ChannelZ-Wk | Still wrong.. the first 1,n,Dial should be 1,1,Dial |
| 21:27.35 | ChannelZ-Wk | s is a different extension. |
| 21:27.39 | swiss__ | ah |
| 21:27.43 | swiss__ | i see |
| 21:28.42 | swiss__ | called using that setup and same thing |
| 21:29.05 | swiss__ | seems like it may not be doing anythign with the digit i input |
| 21:29.42 | swiss__ | ust a guess |
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| 21:32.26 | *** join/#asterisk imox (~imox@p57A1F229.dip0.t-ipconnect.de) |
| 21:32.32 | ChannelZ-Wk | oh... well actually.. I just noticed your console output paste.. |
| 21:33.10 | swiss__ | go on |
| 21:35.18 | ChannelZ-Wk | well your 'findme' isn't even being called from your dialplan. You call the 'screen' macro which reads a digit but then doesn't really do anything. |
| 21:35.50 | swiss__ | ah |
| 21:36.29 | swiss__ | so how do i get that going? |
| 21:37.39 | swiss__ | fyi im basing the extensions.conf on this: http://www.voip-info.org/wiki/view/Asterisk+tips+findme |
| 21:38.08 | *** join/#asterisk insha (~insha@mobile-166-137-177-132.mycingular.net) |
| 21:39.52 | ChannelZ-Wk | Well you just need to combine the concepts |
| 21:41.21 | swiss__ | basically make the findme bit part of screen if im understanding correctly? |
| 21:42.20 | ChannelZ-Wk | IE your multiple dials should be occuring in your [default] context like the first one. |
| 21:42.30 | ChannelZ-Wk | Hang on it's easier to show you |
| 21:42.39 | swiss__ | oki doki |
| 21:43.48 | ChannelZ-Wk | I'm actually a little confused with your setup, in [default] you have exten => _9999XXXXXXXXXX which is presumably what you're dialing to set this off? |
| 21:44.48 | swiss__ | yes, though really that could be just about any number or extension, and again im basing this off the voip-info wiki page i referenced earlier |
| 21:45.27 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 21:46.01 | ChannelZ-Wk | well because your console output doesn't really match the dialplan. So hang on |
| 21:46.38 | swiss__ | hmmm that is odd |
| 21:48.39 | ChannelZ-Wk | that _9999XXXXXXXXXX really your _incoming_ DID or something? |
| 21:49.26 | swiss__ | mope, thats pulled straight from the voip-info wiki page i referenced |
| 21:49.37 | ChannelZ-Wk | well that's why |
| 21:49.42 | swiss__ | i replaced the x's with my number |
| 21:49.47 | swiss__ | in the dialplan |
| 21:49.50 | ChannelZ-Wk | this has no meaning for you |
| 21:49.53 | swiss__ | ha |
| 21:49.55 | swiss__ | ah |
| 21:50.02 | swiss__ | rather and im sorry those x's are ther in the dial plan |
| 21:51.03 | ChannelZ-Wk | http://pastebin.com/PML4ZYUL |
| 21:51.10 | ChannelZ-Wk | here's a testing example./ |
| 21:51.35 | ChannelZ-Wk | Basically you dial 5555 and it's going to dial through those 3 findme numbers (which you need to set) |
| 21:52.38 | ChannelZ-Wk | so you get the concept. I also didn't test it but.. |
| 21:52.53 | ChannelZ-Wk | brb I'm at work and need to wander off |
| 21:53.04 | swiss__ | no wirries, i appreciate the help |
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| 21:55.02 | lvlinux | if i do a Goto(outboundcontext,18883334444,1) and in outboundcontext there is a pattern match _NXXXXXXXXXX, will it go there? |
| 21:57.17 | lvlinux | or will it fail because the actual number 18883334444 isn't found in the context? |
| 21:57.37 | lvlinux | I know Dial() should work but wasn't sure about Goto |
| 21:57.47 | swiss__ | so it looks like the extensions.conf you sent mostly works, though it doesnt seem to grab those phone numbers: |
| 21:57.50 | swiss__ | <PROTECTED> |
| 22:00.42 | swiss__ | ah global should be globals, silly mr |
| 22:12.22 | swiss__ | closer, so now it makes calls and goes through the list but hitting |
| 22:12.32 | swiss__ | 1 doesnt accept the call |
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| 22:22.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:36.37 | KNERD | oooops... wget http://downloads.digium.com/pub/telephony/fax/benchfax/x86-32/\ benchfax-1.1.0-x86_32 -O benchfax |
| 22:36.58 | KNERD | 2014-05-07 17:35:49 ERROR 404: Not Found. |
| 22:37.22 | KNERD | seems the guide needs to be updated |
| 22:37.49 | lanning | remove the "\ " |
| 22:37.56 | KNERD | i did :-) |
| 22:51.22 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 22:52.34 | raspberrypifan | lvlinux r u here |
| 22:52.47 | jmmills | grrg, how do I keep dialplan debugging on but stop these chan_sip handle_response_peerpoke messages |
| 22:52.52 | jmmills | too much chatter! |
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| 23:11.49 | lvlinux | raspberrypifan: yeah i'm here--in and out cuz my connection but i'm here |
| 23:12.06 | lvlinux | 'sup |
| 23:13.43 | raspberrypifan | nm |
| 23:14.08 | raspberrypifan | its about the ata thingie |
| 23:17.07 | [TK]D-Fender | Thingies .... always holding us back... |
| 23:17.20 | lvlinux | yeah? |
| 23:17.29 | raspberrypifan | im gonna ignore [TK]D-Fender |
| 23:17.37 | raspberrypifan | anyway as i was saying |
| 23:17.50 | raspberrypifan | i was told that the ata http://www.ebay.com/itm/131052305172?ssPageName=STRK:MEWNX:IT&_trksid=p3984.m1497.l2649 |
| 23:17.54 | raspberrypifan | would not work |
| 23:17.56 | raspberrypifan | even you had said it would |
| 23:18.44 | lvlinux | well when it comes to * [TK]D-Fender shouldn't be ignored :-D |
| 23:19.00 | lvlinux | what was the reason for it not supposed to work? |
| 23:19.08 | [TK]D-Fender | I would bet that ATA is locked and unusable |
| 23:19.18 | lvlinux | boy that was a badly put sentence... |
| 23:19.23 | raspberrypifan | dk [TK]D-Fender and the other rich fools say it wont work |
| 23:19.28 | raspberrypifan | i told them you said it would |
| 23:19.31 | raspberrypifan | and i trust lvlinux |
| 23:19.33 | raspberrypifan | who ever you are |
| 23:19.41 | lvlinux | well i said IF u get it unlocked it will work. |
| 23:19.56 | lvlinux | the page u found seemed to imply that all of those were relatively easily unlocked. |
| 23:20.09 | raspberrypifan | right but they said that in general |
| 23:20.12 | raspberrypifan | it would not be possible with such an ata |
| 23:20.20 | raspberrypifan | that id neeed all kinds of complex hardware |
| 23:20.58 | [TK]D-Fender | What is the required income level to qualify for your definition of rick exactly? |
| 23:20.58 | lvlinux | like a DES cracker and JTAG interface and such --- you've got all that right???? |
| 23:21.15 | [TK]D-Fender | rich* |
| 23:21.34 | lvlinux | hey [TK]D-Fender I was the first one to tell him to bite the bullet and get a $40 Obi110 FXS/FXO. |
| 23:21.55 | raspberrypifan | he and the other guy said that even the obi would not be enough |
| 23:22.00 | raspberrypifan | that id need a cisco rack unit basically |
| 23:22.04 | [TK]D-Fender | $40 isn't bad for a combo device like that |
| 23:22.23 | [TK]D-Fender | Enough for what? |
| 23:22.28 | lvlinux | a cisco rack unit? to do what? maybe i missed something, but i thot that |
| 23:22.49 | lvlinux | all u were wanting to do is have two numbers in 2 countries and dial to/from either one from anywhere? |
| 23:23.10 | raspberrypifan | yup |
| 23:23.28 | lvlinux | Cisco rack unit???? who told u that? |
| 23:23.55 | lvlinux | or rather, what uber-silly question did u ask that prompted that answer ;-) |
| 23:24.44 | raspberrypifan | i just asked what i asked before, why dont you ask [TK]D-Fender why he said that, it was him and n3glv |
| 23:25.09 | [TK]D-Fender | You're the one quoting what you were supposedly told. |
| 23:25.37 | [TK]D-Fender | So if you're going quote don't forget attribution |
| 23:26.42 | [TK]D-Fender | And I can confirm I never mentioned "rack" anywhere for anything in weeks, let alone something like this |
| 23:27.31 | [TK]D-Fender | raspberrypifan: So rather than wasting time, I will give you a direct link to the minimal equipement I would personally recommend if you will clearly break down what you are looking to connect to. |
| 23:30.49 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-169.ph.ph.cox.net) |
| 23:32.58 | lvlinux | raspberrypifan: u sure ur not in another channel and getting this info? :-D |
| 23:33.15 | raspberrypifan | i was in #voipcoop but it had [TK]D-Fender in it |
| 23:33.51 | [TK]D-Fender | raspberrypifan: Do you want a staght and simple solution to your need or not? |
| 23:33.55 | [TK]D-Fender | straight* |
| 23:37.38 | raspberrypifan | is the jtag thing the one with 4 pin thingis |
| 23:38.34 | lvlinux | raspberrypifan: IF you can get that unit unlocked then it will allow u to connect to ur magicjack. There's ur US number. Then for the Ecuador side u want to either get another FXO device and a landline or a SIP DID. |
| 23:39.08 | lvlinux | raspberrypifan: that was a joke--if u have to do all that to unlock an ATA, it's not worth it, just spend a few bucks more. |
| 23:39.09 | raspberrypifan | there are no SIP DIDs i looked, so i would have a landline |
| 23:39.16 | raspberrypifan | to me it is |
| 23:39.21 | raspberrypifan | remember i got the time not the money |
| 23:39.34 | lvlinux | trust me u don't have that much time. |
| 23:39.49 | *** join/#asterisk haroldp (~Digger@99-46-25-48.lightspeed.renonv.sbcglobal.net) |
| 23:39.59 | raspberrypifan | i do |
| 23:40.03 | raspberrypifan | ive had to do jtag |
| 23:40.04 | raspberrypifan | for routers before |
| 23:40.15 | lvlinux | and if u do, you can spend it making tons of money putting to use the knowlege of that stuff for pay jobs. |
| 23:40.18 | raspberrypifan | i screwed up on the soldering |
| 23:40.24 | raspberrypifan | i dont like to work |
| 23:40.26 | raspberrypifan | for companies |
| 23:40.27 | raspberrypifan | im against it |
| 23:40.41 | lvlinux | i'ts not just jtag---you'd have to crack codes and reengineer firware etc. |
| 23:40.47 | lvlinux | it's just silly. |
| 23:40.53 | raspberrypifan | well...i cant do those two other tings |
| 23:40.57 | lvlinux | so don't work for a company freelance. |
| 23:40.57 | raspberrypifan | i dont have the knowledge |
| 23:41.09 | raspberrypifan | im against capitalism, the thing i do, i do for my enjoyment |
| 23:41.12 | raspberrypifan | and my enrichement |
| 23:41.15 | raspberrypifan | on a personal level |
| 23:41.38 | lvlinux | ok without going into that, lets get back to what you need to do. |
| 23:41.44 | raspberrypifan | right |
| 23:41.45 | [TK]D-Fender | Looks like no point in helping here.... |
| 23:41.47 | [TK]D-Fender | moves on. |
| 23:41.49 | raspberrypifan | so what i need two ata is what u saying? |
| 23:41.56 | RaMcHiP | lol |
| 23:42.08 | lvlinux | two fxo interfaces. |
| 23:42.22 | *** join/#asterisk Dovid (~Dovid@ool-4a584bd2.dyn.optonline.net) |
| 23:42.35 | raspberrypifan | is that two line ports? |
| 23:42.35 | lvlinux | or the other solution is to get the SIP credentials from the MJ and use them directly in ur Asterisk on pi. |
| 23:42.49 | raspberrypifan | theyve banned u getting the sip stuff |
| 23:43.11 | lvlinux | oh you've done it? |
| 23:44.01 | raspberrypifan | well |
| 23:44.05 | raspberrypifan | i read about a bunch of times |
| 23:46.34 | *** join/#asterisk Alex_Bkash (~atique@124.6.236.252) |
| 23:53.59 | haroldp | So I can receive calls but not place them through my asterisk server. My voip provider always returns a 403 unauthorized. |
| 23:55.39 | haroldp | talking to their support, they told me that the FROM header in my INVITE looks wrong. I’m sending something like FROM: <sip:username@my.ip.address> and they insist it should be FROM: <sip:username@their.pop.server> |
| 23:55.49 | haroldp | does that seem right? |
| 23:56.06 | [TK]D-Fender | haroldp: fromdomain=their.domain.name.here |
| 23:56.27 | haroldp | in the general section of sip.conf? |
| 23:56.52 | [TK]D-Fender | no, in you rpeer you use to dial them |
| 23:57.18 | haroldp | aha |
| 23:57.39 | haroldp | er.. |
| 23:58.32 | haroldp | so like: register => me@there.dom.ain:secret@their.dom.ain |
| 23:58.37 | [TK]D-Fender | no. |
| 23:58.45 | [TK]D-Fender | I did not say in your register statement. |
| 23:58.48 | [TK]D-Fender | I said in your PEER |
