| 00:19.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:19.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 00:39.08 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:46.00 | Katty | looks in |
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| 00:53.22 | carrar | looks out |
| 00:54.23 | Katty | shakes it all about? |
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| 02:00.41 | *** join/#asterisk m78pl (32c2ec51@gateway/web/freenode/ip.50.194.236.81) |
| 02:00.52 | m78pl | anyone in the hotel in atlanta ? |
| 02:01.51 | m78pl | noone going for astricon ? |
| 02:01.57 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 02:02.24 | volga629 | Hello Everyone. from where start look WARNING[25048] chan_sip.c: Failure to write to tcp/tls socket |
| 02:02.35 | volga629 | what possible cause |
| 02:05.47 | m78pl | is it just tcp or ssl ? |
| 02:05.59 | volga629 | TLS client |
| 02:06.08 | volga629 | look like lost connection |
| 02:06.12 | m78pl | so secure ... |
| 02:06.16 | volga629 | yes |
| 02:06.23 | m78pl | yes, it looks like write() cannot write to the file descriptor |
| 02:06.44 | volga629 | which file ? might permission problem |
| 02:07.03 | m78pl | tcptls_session->fd |
| 02:08.33 | m78pl | is this on incoming calls ? |
| 02:08.40 | volga629 | yes |
| 02:08.47 | volga629 | user dial in |
| 02:09.24 | RaNa | hey guys im trying to setup time conditions based on weeks but i cant seem to find a way to setup change ivr every week |
| 02:09.27 | m78pl | or the code can't bind to the proper ip address ... |
| 02:09.36 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 02:09.58 | m78pl | rana: to setup change ivr every week ? what do you mean ? |
| 02:11.05 | RaNa | i have need a diff msg every week i want to where every week it changes the msg |
| 02:11.25 | Penguin | You have to write that in dial plan. |
| 02:11.27 | RaNa | i have 2 diffrent msg that i need to play every othere week |
| 02:11.34 | RaNa | i hope that make sense |
| 02:11.35 | m78pl | volga: but if it's secure then maybe you don't have keys ... |
| 02:12.57 | *** join/#asterisk imcdona (~Thunderbi@gateway.defiancedc.com) |
| 02:17.12 | redotis | Whenever I use SayNumber my phone says declined? |
| 02:18.57 | volga629 | checking all possible, but this client was connected for last 2 weeks |
| 02:19.23 | m78pl | hmmm, what has changed them |
| 02:19.53 | volga629 | possible want login to desktop and check the client |
| 02:19.58 | m78pl | redotis: probably your phone is hitting wrong context ... and asterisk is not executing anything |
| 02:20.43 | m78pl | rana: i guess it's complicated but you can you strftime, print the week of the year then % 2 and you have 0 and 1 always ... |
| 02:20.47 | redotis | Thanks m78pl...it's is in the right context. The extension is executed |
| 02:21.02 | m78pl | then just prepare the two prompts prompt_1 and prompt_0 |
| 02:22.20 | m78pl | rana: http://www.voip-info.org/wiki/view/Asterisk+func+strftime |
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| 02:23.38 | RaNa | thanx m78pl ill take a look atit |
| 02:23.41 | RaNa | it |
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| 02:26.00 | m78pl | anyone in Atlanta right now ? |
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| 02:34.54 | volga629 | on server time offf |
| 02:35.11 | volga629 | ntpd screw up |
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| 02:37.39 | m78pl | interesting |
| 02:38.02 | m78pl | volga ... is your ntpd not syncing right ? |
| 02:38.16 | volga629 | yes, correcting right now |
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| 04:23.12 | *** join/#asterisk mae_tae (ca4e5973@gateway/web/freenode/ip.202.78.89.115) |
| 04:23.19 | mae_tae | hello people |
| 04:23.23 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:23.25 | mae_tae | it seems that snort is blocking my call outside the campus, how do i fix this? if i stop the snort, everything is working fine |
| 04:29.06 | [TK]D-Fender | Perhaps configureing snort to keep out of the way.... |
| 04:33.16 | mae_tae | yeah, how do i exempt or allow external from snort |
| 04:38.01 | [TK]D-Fender | mae_tae: that's a question better suited to a snort support channel.... |
| 04:38.20 | mae_tae | oh yeah...... ok |
| 04:39.53 | redotis | What's this all about? sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead |
| 04:40.47 | Penguin | looks around |
| 04:42.15 | [TK]D-Fender | looks asquare |
| 04:42.36 | Penguin | That's odd... |
| 04:42.45 | Penguin | I always took you for atriangle. |
| 04:43.30 | mae_tae | i find it arectangle |
| 04:43.36 | Penguin | I don't know whether to laugh or feel sorry for that guy. |
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| 05:28.59 | jeev | if i have changed bindport to 4554 for example for iax, and in trunk configs have port=4554, two trunks are fine but the one i just added is still trying 4569, what causes this ? |
| 05:33.03 | *** join/#asterisk jhlavacek (~jirka@80.215.0.197) |
| 05:34.06 | jeev | iax2 show peers shows it at 4554 but it's attempting 4569. |
| 05:34.08 | [TK]D-Fender | jeev: You are unclear about what is set where on which end |
| 05:35.11 | stanlly | I'm having issues with wcte11xp, I'm only seeing outgoing pri frames nothing comes back in paste http://pastebin.com/N3zd0HrZ any ideas? should I change the versions of dahdi and libpri for asterisk 1.4.44 ? |
| 05:37.30 | jeev | fender, host=ip and port=4554 on the next line, along with bindport beint set to 4554. just how my other trunks are working on 4554 right now, why is this one showing 4554 in iax2 show peers but iax2 set debug shows it trying to communicate on 4569 |
| 05:38.53 | kaldemar | what is trying to communicate and in which direction? |
| 05:39.13 | jeev | office A is talking to B, B talking to A, they are just hitting unreachable. |
| 05:39.59 | kaldemar | and how are we supposed to know what you have configured in A and what in B? |
| 05:40.27 | jeev | kaldemar, host= and port=. |
| 05:40.30 | [TK]D-Fender | [01:34][TK]D-Fenderjeev: You are unclear about what is set where on which end |
| 05:40.35 | [TK]D-Fender | jeev: WHAT ON WHAT SIDE |
| 05:40.39 | jeev | fender, i always thought i dont understand you |
| 05:40.40 | jeev | BOTH SIDES. |
| 05:40.46 | jeev | both have the same config, differnet ips |
| 05:40.51 | jeev | just how the OTHER DAMN TRUNKS WORK |
| 05:40.52 | [TK]D-Fender | show us the configs |
| 05:41.01 | [TK]D-Fender | your description is terribly vague |
| 05:41.21 | [TK]D-Fender | And the CLI status dumps while you're at it |
| 05:54.42 | jeev | omfg. |
| 05:54.49 | jeev | does anyone here have a gun |
| 05:54.53 | jeev | please use it on me now |
| 05:55.56 | [TK]D-Fender | please stand on that large plastic sheet in the corner and we'll get to you in the order in which your call was received. |
| 05:56.03 | [TK]D-Fender | starts MoH |
| 05:56.09 | jeev | do you know what it was? |
| 05:56.30 | jeev | iptables. |
| 05:56.37 | [TK]D-Fender | Almost assuredly a 1D-10-T error :) |
| 05:56.44 | jeev | both servers only allow specific ips to that port. |
| 05:57.30 | jeev | i have such poor memory |
| 05:58.55 | jeev | fender, i know it's not recommended to change iax2 port but if i did.. registration is still sending to default port, do i simplly add :4554 at the end ? |
| 05:59.15 | jeev | in the scenario where they're both public IP.. i dont need to really register, right ? |
| 05:59.51 | kaldemar | it has more to do with the addresses being non-dynamic. |
| 06:00.29 | [TK]D-Fender | ^ |
| 06:00.57 | [TK]D-Fender | Non-moving target |
| 06:17.55 | [TK]D-Fender | Checkout time, later all.... |
| 06:23.05 | jeev | kaldemar, not trying to jinx myself.. |
| 06:23.16 | jeev | when directly connecting via iax2, it kept going unreachable. |
| 06:23.25 | jeev | when running just a basic ptp openvpn, it does not go unreachable. |
| 06:23.47 | jeev | would this be an MTU issue since openvpn handles the MTU itself ? |
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| 08:38.22 | *** join/#asterisk uosiu (~uosiu@hajdamowicz.info) |
| 08:38.25 | uosiu | Hi all |
| 08:39.01 | uosiu | One of agents inside company gave me an info that he can't call phones outside our company |
| 08:39.21 | uosiu | Asterisk (core set verbose=99) gave me this log: http://fpaste.org/45023/81221448/ |
| 08:39.34 | uosiu | Is it problem at our site or on remote site? |
| 08:40.04 | uosiu | We use SIP internally and on our remote provider. |
| 08:41.14 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:41.46 | mirela666 | well extension 31 and tvk provider were bridged |
| 08:41.47 | mirela666 | Locally bridging SIP/31-00002d95 and SIP/tvk-00002d96 |
| 08:43.02 | mirela666 | uosiu: what is the time difference between Localy bridging and Spawn extension, the next step |
| 08:43.23 | mirela666 | uosiu: how long were they connected? |
| 08:48.34 | *** join/#asterisk andrewyager (~andrewyag@101.168.50.139) |
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| 08:56.25 | *** join/#asterisk galba (~pif@84-74-26-25.dclient.hispeed.ch) |
| 08:57.16 | galba | hi, why did asterisk go from 1.8.* to 11.5.* versions ? |
| 08:57.43 | galba | also, how do I lookup a function at the cli (e.g Dial)? |
| 08:58.08 | mirela666 | galba: core show application Dial |
| 08:58.13 | galba | ah yes |
| 08:58.28 | mirela666 | galba: and core show functions |
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| 10:51.18 | gartral | where do I find the dependency "iksemel" |
| 10:53.43 | kaldemar | gartral: which OS are you on? |
| 10:54.14 | gartral | kaldemar: nvm, derp, (E) stands for external, and for future reference, ubuntu |
| 10:55.01 | gartral | kaldemar: i'm probably slightly too drunk to be working on compiling asterisk. But I want working gvoice, damnit |
| 10:57.15 | gartral | libiksemel-dev is installed, but isn't being read by asterisk.. should I make clean and reconfigure? |
| 10:59.16 | gartral | that did it |
| 10:59.52 | kaldemar | re-running the configure script is required after you install dependencies. |
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| 11:03.34 | gartral | kaldemar: yea, it's been a while since I've build anything from sources >.> |
| 11:11.18 | gartral | should i turn on any embedding? |
| 11:12.04 | Greenlight | I'm using cdr_adaptive_odbc to insert CDR's to two separate db's, the idea being that if one fails, we still have CDR's going to the other and won't loose any. However, it seems that if one database becomes unavailable the CDR's going into the other become sporadic (i'm seeing CDR's inserted 2 hours *after* the call ends)... anyone have any ideas what might be happening here ? |
| 11:12.39 | kaldemar | gartral: if you need to ask, the answer is probably no. |
| 11:13.47 | gartral | kaldemar: ok, smarter question: what do those options do? |
| 11:16.46 | kaldemar | they embed functionality into the asterisk binary instead of compiling them into separate modules. |
| 11:35.36 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
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| 11:51.37 | Greenlight | Hmm... even my raw cdr.csv files the CDRs are completely out of order... a call ending at 9:28 followed by an entry cal a call ending at 8:53 |
| 11:53.13 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:54.32 | Ice_Strike | Hey Greenlight |
| 11:54.41 | Greenlight | Afternoon Ice_Strike |
| 12:00.45 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:08.16 | Ice_Strike | AMD() is being an ass |
| 12:08.49 | Ice_Strike | Greenlight do you use AMD() ? |
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| 12:12.24 | Greenlight | We have a couple of methods to use it |
| 12:12.41 | Greenlight | You shuld try playback of a very very short silence before going to AMD though |
| 12:12.54 | Greenlight | This helps to ensure that the media is "up" etc before you hit amd |
| 12:13.21 | Ice_Strike | Greenlight Which playback |
| 12:13.32 | Greenlight | Just make a short file of silence |
| 12:13.53 | Ice_Strike | what do I save it as |
| 12:13.53 | file | Playback(silence/1) |
| 12:14.07 | Ice_Strike | @file isnt /1 1 second |
| 12:14.11 | Greenlight | One second may be too long |
| 12:14.11 | Ice_Strike | might be too much |
| 12:14.15 | Ice_Strike | yea |
| 12:14.24 | Greenlight | Just chop it down though |
| 12:14.40 | Ice_Strike | I use Audacity |
| 12:14.45 | file | true! |
| 12:14.48 | Ice_Strike | save as wav? |
| 12:14.54 | Greenlight | Yea |
| 12:15.01 | Greenlight | 100ms or something |
| 12:15.18 | Greenlight | WE generaly don't use AMD that way cause of false positives |
| 12:15.35 | Ice_Strike | I have replaced from old router to a new router at work... I am getting alot of false positives via AMD |
| 12:15.42 | Ice_Strike | it keep returning Machine |
| 12:15.50 | Ice_Strike | Never had problem with old router |
| 12:15.58 | Greenlight | Umm.. how can the router effect your AMD? |
| 12:16.13 | Ice_Strike | That is why I don't understand |
| 12:16.29 | Ice_Strike | I tried switchign back to old router and it work fine. |
| 12:16.41 | Greenlight | That's.... strange |
| 12:17.11 | Greenlight | Is it perhaps causing "no audio" type issues or something, which AMD is detecting as Machine ? |
| 12:17.20 | file | sounds (ha!) like it |
| 12:17.30 | Greenlight | ^^ |
| 12:18.47 | Ice_Strike | Greenlight I have also notice when orginating the calls via AMI - when the user answer it - it did not execute the dial plan |
| 12:18.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:19.19 | Greenlight | Ice_Strike: That makse no sense... if the user answers then it gets dropped into the dialplan.. |
| 12:19.27 | Greenlight | Sounds like something funky is going on |
| 12:19.31 | Ice_Strike | Yea it should |
| 12:19.35 | Ice_Strike | I know :/ |
| 12:19.45 | Ice_Strike | maybe sip debug might help? |
| 12:20.07 | Ice_Strike | I mean figure out the problem via debug |
| 12:20.21 | Ice_Strike | I will try pastebin it and if you can see the problem |
| 12:20.32 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 12:20.42 | Greenlight | Sounds very much like a network issue |
| 12:28.06 | Greenlight | Can anyone tell me if "negative_connection_cache", a configuration parameter in cdr_adaptive_odbc.conf has a default value, or even if it's enabled by default? |
| 12:32.55 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
| 12:39.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:39.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 12:58.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 12:59.07 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 13:02.02 | volga629 | Hello Everyone, I am getting this warning |
| 13:02.05 | volga629 | <PROTECTED> |
| 13:02.07 | volga629 | [2013-10-08 14:56:32] WARNING[10466]: translate.c:1068 __ast_register_translator: Table cost could not be generated for g729tolin, Please set table_cost variable on translator. |
| 13:02.25 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 13:03.51 | pabelanger | volga629: contact digium support |
| 13:04.10 | *** part/#asterisk pigeonflight (~macuser@72.252.224.80) |
| 13:05.45 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
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| 13:09.06 | Katty | morning |
| 13:13.33 | jmetro | mernin |
| 13:15.49 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 13:15.50 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 13:16.15 | Ice_Strike | I keep getting this msg: |
| 13:16.16 | Ice_Strike | <PROTECTED> |
| 13:16.17 | Ice_Strike | <PROTECTED> |
| 13:16.56 | Ice_Strike | in the dial plan its: exten => s,1(Box),Answer() |
| 13:17.34 | BorjaGVO | Hi everyone. Suddenly, Asterisk stopped without me doing anything (that I know). Luckly, I had the Console opened and this is what I found: http://pastebin.com/56K1WusC |
| 13:17.46 | BorjaGVO | Why did this happen? :-/ |
| 13:18.17 | *** join/#asterisk RaNa (~WinNT@97-71-95-51.res.bhn.net) |
| 13:18.46 | Greenlight | BorjaGVO: YOu had a seg fault |
| 13:18.57 | Greenlight | Which version of asterisk are you using ? |
| 13:20.02 | BorjaGVO | Greenlight: 1.8.23.0 |
| 13:20.05 | [TK]D-Fender | Ice_Strike: Clearly not in the right place |
| 13:20.12 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:20.22 | Greenlight | In that case, https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 13:20.26 | [TK]D-Fender | Ice_Strike: and you shouldn't be failing the first thing it's looking for.... |
| 13:20.41 | Greenlight | Get a backtrace, and raise an issue on the JIRA |
| 13:20.48 | Ice_Strike | [TK]D-Fender Problem caused when changed a router |
| 13:21.16 | BorjaGVO | Greenlight: Alright |
| 13:21.21 | Greenlight | Ice_Strike: You should fix your network issues first |
| 13:21.37 | Ice_Strike | I am trying to figure out. |
| 13:22.25 | *** join/#asterisk tzafrir (~tzafrir@63.133.202.2) |
| 13:22.27 | Ice_Strike | how can I show you a debug sip? |
| 13:23.11 | [TK]D-Fender | PASTEBIN <- |
| 13:23.37 | Katty | refills jmetro's coffee |
| 13:23.47 | jmetro | yerba mate, plz |
| 13:24.02 | Katty | statement does not parse. please update indexes and try again. |
| 13:32.12 | Ice_Strike | [TK]D-Fender may I send you a pastebin link via PM? |
| 13:33.15 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 13:33.49 | qakhan | hi all, how i can check number of calls in a queue, |
| 13:34.19 | [TK]D-Fender | Ice_Strike: yes |
| 13:34.28 | [TK]D-Fender | qakhan: "queue show" |
| 13:34.40 | kaldemar | qakhan: tab is your friend in CLI. |
| 13:34.44 | qakhan | i want to move thrid caller in a queue to automate system |
| 13:36.25 | qakhan | guys i am taking about dial plan |
| 13:37.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:37.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:37.14 | [TK]D-Fender | qakhan: No. Hijacking a call out of a queue is beyond dialplan. This requires AMI <- |
| 13:37.25 | Katty | waves at sruffell |
| 13:37.29 | sruffell | hi Katty |
| 13:37.35 | Katty | how're you dear |
| 13:38.08 | sruffell | Pretty good. How about yourself? |
| 13:38.21 | Katty | good good. bit sleepy still (= |
| 13:38.50 | *** join/#asterisk mjordan (~mjordan@63.133.202.2) |
| 13:38.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:39.09 | [TK]D-Fender | Ice_Strike: So what's the problem in there? |
| 13:39.22 | Katty | waves to mjordan |
| 13:40.02 | kaldemar | qakhan: "core show functions like QUEUE" |
| 13:40.18 | [TK]D-Fender | Ice_Strike: Looks like an Originate call-out. And your NAT setup is WRONG |
| 13:41.09 | Ice_Strike | [TK]D-Fender When I changed to a new router - when orginating the calls via AMI - it did not excute the dial plan |
| 13:41.21 | Ice_Strike | Not showing any problem on log? |
| 13:41.27 | Ice_Strike | Oh |
| 13:42.53 | [TK]D-Fender | Ice_Strike: I wouldn't be surprised at a rejection at any pointin the call including 1-way audio at best |
| 13:43.05 | Ice_Strike | http://pastebin.com/bqt07vU5 |
| 13:43.10 | Ice_Strike | That is what I have |
| 13:43.22 | [TK]D-Fender | I also see dialplan errors being reported |
| 13:44.01 | [TK]D-Fender | Ice_Strike: Ice_Strike that peer is wrong as is "callmanager" |
| 13:44.06 | [TK]D-Fender | Ice_Strike: And your core NAT settings |
| 13:44.39 | qakhan | kaldemar i cannot find any reasonable information |
| 13:44.55 | [TK]D-Fender | qakhan: "queue show" tells you who is in the queue |
| 13:45.05 | [TK]D-Fender | qakhan: And you have not speicified exactly what you are looking for |
| 13:45.20 | [TK]D-Fender | specified |
| 13:46.05 | qakhan | [TK]D-Fender i told you |
| 13:46.16 | Ice_Strike | [TK]D-Fender I am getting a big confused now. What do I need to change in the sip.conf that I show u |
| 13:46.22 | qakhan | i want to move thrid caller in a queue to automate system |
| 13:47.00 | qakhan | or you can say i want to move third caller to supervisor ext |
| 13:47.01 | [TK]D-Fender | Ice_Strike: a lot. all of the [general] settings to work from behind NAT, and that NONE of those peers are behind NAT and you hsould have them all as "nat=no" |
| 13:47.47 | Ice_Strike | Ah |
| 13:47.51 | [TK]D-Fender | [09:33]qakhanhi all, how i can check number of calls in a queue, [09:34]qakhani want to move thrid caller in a queue to automate system [09:37][TK]D-Fenderqakhan: No. Hijacking a call out of a queue is beyond dialplan. This requires AMI <- |
| 13:48.05 | Ice_Strike | [TK]D-Fender I will give a try later. |
| 13:48.12 | [TK]D-Fender | qakhan: Hijack = AMI <- |
| 13:48.42 | [TK]D-Fender | Ice_Strike: And for the other side... your dialplan errors are what they are.... you don't have it in the right place |
| 13:49.40 | mirela666 | Hey all, |
| 13:49.59 | mirela666 | Can out of band DTMF aka rfc2833 detect tones as DTMF ? |
| 13:50.26 | [TK]D-Fender | OB directly means that it doesn't "detect". It is told. |
| 13:50.28 | [TK]D-Fender | OOB |
| 13:51.16 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-69-214.user.veloxzone.com.br) |
| 13:51.41 | mirela666 | I'm having some wierd thing, all sip peers have rfc2833 but when I turn sound to max on keypad it starts detecting digits twice |
| 13:54.47 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 13:55.05 | [TK]D-Fender | What keypad? please describe the complete call... |
| 13:55.29 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:56.10 | mirela666 | snom300 phone keypad |
| 13:56.18 | anonymouz666 | nice setup... 2 hosts 2 guests running 4 KVM and four corosync nodes |
| 13:56.26 | [TK]D-Fender | And the other side? |
| 13:56.35 | mirela666 | all extensions have rfc2833 |
| 13:56.49 | [TK]D-Fender | mirela666: Please be specific with this question..... |
| 13:57.05 | mirela666 | but in general in sip.conf I have dtmfmode=inband |
| 13:57.28 | mirela666 | DTMF is doubling sometimes |
| 13:57.47 | mirela666 | instead of 1234 i get in log 123444 |
| 13:57.51 | mirela666 | for example |
| 13:59.20 | mirela666 | it's not the pphone, it's the setup that was before I started working on I can't figure it out at all |
| 13:59.48 | mirela666 | sip show peer XXX shows dtmfmode as rfc2833 |
| 14:00.08 | mirela666 | but I guess the general section is overwriting it |
| 14:00.20 | [TK]D-Fender | it doesn't |
| 14:00.27 | mirela666 | if I coment the general section DTMF is not recognised |
| 14:00.45 | mirela666 | or if I chqange it to rfc2833 |
| 14:02.12 | *** join/#asterisk mjordan (~mjordan@63.133.202.2) |
| 14:02.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:26.30 | *** part/#asterisk RaNa (~WinNT@97-71-95-51.res.bhn.net) |
| 14:26.32 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
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| 14:52.44 | *** join/#asterisk Sparky-UK (~not@host-80-47-239-28.as13285.net) |
| 14:53.39 | Sparky-UK | Hi, can anyone tell me if Asterisk can be used to issue access ISDN level protocol, or is this something that is just completely abstracted by Asterisk? |
| 14:55.16 | roswell | Sparky-UK, you will need an isdn/sip gateway (or isdn board) |
| 14:56.15 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 14:56.18 | Sparky-UK | I am not refering to SIP here, I am on about the ISDN side, there are certian network level queires that I need to perform, but I dont know if asterisk can do that |
| 14:56.59 | Sparky-UK | Can asterisk do that? or is there another way to interface with the ISDN board directly? |
| 15:22.41 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
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| 15:38.24 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 15:42.52 | ickmund | Hmm... with endbeforehexten=yes, should I still be able to change fields in the CDR in the h extension? |
| 15:51.18 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 15:54.10 | davlefouAMD | hi, how i made an sip register with realtime? I have my sipregs on sip data base |
| 16:03.40 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:09.11 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 16:12.41 | [TK]D-Fender | [10:56]Sparky-UKI am not refering to SIP here, I am on about the ISDN side, there are certian network level queires that I need to perform, but I dont know if asterisk can do that <- no. |
| 16:13.00 | [TK]D-Fender | Sparky-UK: You can see standard call attempts, etc, but for real debugging, no |
| 16:13.49 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 16:16.01 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 16:17.47 | jeev | fender |
| 16:17.51 | jeev | buddy.. pal.. |
| 16:18.13 | jeev | brother.. |
| 16:18.40 | jmetro | brosephone |
| 16:18.42 | jeev | you remember my constant complaints recently about the iax2 peers just going unreachable constantly? |
| 16:22.54 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:23.28 | ChannelZ | brohan |
| 16:25.23 | jeev | with iax2 public ip to public ip, it would go unreachable every minute. when i ptp openvpn them together, it has been stable the whole night |
| 16:26.28 | Greenlight | Any NAT involved ? |
| 16:26.42 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
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| 16:27.40 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 17:22.24 | qakhan | can we get queue stat from other asterisk like this |
| 17:22.26 | qakhan | exten => 5501,1,Set(qc=IAX2/regencysip/${QUEUE_WAITING_COUNT(regencytaxi)}) |
| 17:22.27 | qakhan | ? |
| 17:22.53 | roswell | no |
| 17:23.36 | *** join/#asterisk bmg505 (~leon@196-209-99-92.dynamic.isadsl.co.za) |
| 17:24.03 | jeev | Greenlight, well, public ip, no nat.. |
| 17:24.39 | roswell | qakhan, though these guys seem like finding a solution http://forums.digium.com/viewtopic.php?p=63327 |
| 17:28.11 | Ice_Strike | [TK]D-Fender thank you very much - disabling the NAT fixed the problem and also added Playback(silence/1) before AMD() as a test.. work great |
| 17:28.35 | Ice_Strike | now im going to make like 0.1 sec audio file somehow. |
| 17:29.57 | Penguin | Why? |
| 17:30.19 | Penguin | Do you simply need to answer the line and then wait 100ms before AMD starts? |
| 17:31.00 | *** join/#asterisk serafie (~erin@nat/digium/x-luxnoquergtpgvvw) |
| 17:31.13 | Ice_Strike | Penguin Yes like that |
| 17:31.16 | Penguin | Answer(100) before AMD() |
| 17:31.38 | Ice_Strike | Penguin AMD reported a lot of machine which is false. |
| 17:31.49 | Ice_Strike | so I have tried Playback(silence/1) and it work great |
| 17:31.56 | Ice_Strike | Hmm Answer(100) |
| 17:32.08 | Penguin | Do you have something before the Playback() that answers the line? |
| 17:32.43 | Ice_Strike | First: Answer(), 2 UserEvent(), and then AMD() |
| 17:33.06 | Ice_Strike | So I added Playback() after Answer() and it work |
| 17:33.08 | Penguin | And you put the Playback() between the UserEvent and the AMD? |
| 17:33.11 | [TK]D-Fender | [13:31]PenguinAnswer(100) before AMD() <- that is a wait after answer, but doesn't force audio. |
| 17:33.22 | [TK]D-Fender | Penguin: You really want to guarantee RTP setup... |
| 17:33.28 | Penguin | Answer() starts RTP, doesn't it? |
| 17:33.33 | [TK]D-Fender | not really... |
| 17:33.37 | Penguin | It's supposed to. |
| 17:33.49 | [TK]D-Fender | not forcible in the standard... |
| 17:33.59 | [TK]D-Fender | You'd think so, but hey... |
| 17:34.09 | Penguin | So if you Answer and make answer wait for 100ms after it answers, it should be good. |
| 17:34.26 | [TK]D-Fender | nope, need actual audio to gurantee the setup |
| 17:34.38 | [TK]D-Fender | stupid delay could add just enough to make AMD suck |
| 17:34.44 | Ice_Strike | What shall I save as in Audacity? |
| 17:34.48 | Penguin | Answer(100) would be the same as Answer() Wait(0.1) |
| 17:35.01 | Penguin | mono, 8-bit |
| 17:35.11 | Penguin | Wait, that's not right. |
| 17:35.23 | ChannelZ-Wk | mono 16-bit 8khz |
| 17:35.33 | Penguin | There! |
| 17:35.36 | Penguin | I knew it wasn't right when I typed it. |
| 17:36.12 | ChannelZ-Wk | although I never understod the 16-bit part, isn't ulaw 8? |
| 17:36.48 | Penguin | I don't think it's related to ulaw in any way. |
| 17:37.32 | drmessano | 8-bit sampling would sound horrible lol |
| 17:37.33 | ChannelZ-Wk | well it's not going to get any better. But I guess it's probably a math thing of the encoding, maybe it's helpful to have more data |
| 17:37.47 | ChannelZ-Wk | I ain't no professor |
| 17:37.51 | Penguin | I knew I said the wrong thing as soon as I pressed enter. |
| 17:38.25 | ChannelZ-Wk | I do that a lot |
| 17:38.52 | *** join/#asterisk newtonr (~newtonr@63.133.202.2) |
| 17:38.52 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:38.57 | Penguin | It's crappy that backspace doesn't work after pressing enter. |
| 17:39.11 | ChannelZ-Wk | I thought about creating a fake mailing list for myself, because it always seems like when I'm having a problem, I finally give up and post a message somewhere about it, and then 2 minutes later I figure it out |
| 17:39.26 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:39.40 | drmessano | Text yourself |
| 17:39.57 | drmessano | The arguments are even more difficult in single-player notepad |
| 17:40.06 | drmessano | Multi-player Notepad is easy |
| 17:40.31 | drmessano | If you text yourself and it doesn't escalate quickly, you're doing it wrong |
| 17:43.50 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 17:49.54 | Ice_Strike | Where do I set 8khz in Audacity? |
| 17:50.50 | navaismo | project rate |
| 17:51.17 | navaismo | or track-->resample |
| 17:51.39 | Penguin | Look in the lower left corner. |
| 17:52.29 | Ice_Strike | Ah |
| 17:52.30 | Ice_Strike | thnaks |
| 17:53.00 | roswell | or rather you may choose 'other uncompressed formats' when exporting, hit 'parameters' and set the appropriate |
| 17:54.38 | Ice_Strike | where is 'other uncompressed formats' |
| 17:55.53 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 17:56.16 | roswell | file -> export (or, export selection [if any]), choose the first filetype from the dropdown |
| 17:57.44 | roswell | specify 'raw' for the header, and set the preferred format |
| 17:59.30 | roswell | meh... 'other uncompressed files' that is heh |
| 17:59.56 | qakhan | roswell mentioned link didnt help me |
| 18:00.19 | qakhan | there is any way we can get veriable value from other asterisk server |
| 18:01.41 | [TK]D-Fender | qakhan: get it when? |
| 18:01.57 | roswell | qakhan, i very doubt that. well try passing in an unused callerid parameter |
| 18:02.05 | kaldemar | soon |
| 18:02.09 | Ice_Strike | meh when i set to 8khz its coverted to like 5 mins |
| 18:02.22 | roswell | [TK]D-Fender, he's talking about passing an arbitrary parameters between * servers |
| 18:02.37 | [TK]D-Fender | Passing is different than getting. |
| 18:02.42 | kaldemar | both iax and sip offer ways to carry information. |
| 18:02.46 | [TK]D-Fender | More people with broken descriptions |
| 18:04.05 | Ice_Strike | roswell cant get it to work, can you do me a favor? :P |
| 18:04.08 | navaismo | i want a sandwich |
| 18:04.20 | Ice_Strike | create 100ms blank file |
| 18:04.43 | navaismo | uh? with the sandwich? |
| 18:05.42 | Penguin | Take a copy of silence/1.wav into audacity. Cut off 900ms of it. Save as a new file name. |
| 18:05.53 | roswell | Ice_Strike, share the file somewhere and drop me a link |
| 18:06.03 | Penguin | Stop complicating things. |
| 18:06.05 | qakhan | ok i have 2 asterisk servers. server A get calls and send them to queue. server B is for TTS. Server B takes caller id and search in DB and get record and speack to callers. |
| 18:06.16 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 18:07.04 | navaismo | Use SOX |
| 18:07.34 | qakhan | here server B need to check QUEUE_MEMBER_COUNT(). if there are 2 callers in the queue then call send to Server A. |
| 18:08.48 | roswell | qakhan, if so, you only have to set the appropriate callerid - Set(CALLERID(1..)=${2..}) - before Dial() in your dialplan |
| 18:13.12 | [TK]D-Fender | Pass it in the callerID. Or pass it in a SIP header |
| 18:13.21 | roswell | where 1.. is an appropriate cid field, and 2.. is a variable |
| 18:13.22 | [TK]D-Fender | And calling BACK to B loooks wasteful. |
| 18:13.51 | kaldemar | caller id does not appreciate being molested like that. |
| 18:14.13 | [TK]D-Fender | We also see you saying to send the call back... but not what to do in the OTHER case |
| 18:14.36 | [TK]D-Fender | Descriptions a re still very bad from everyone today |
| 18:17.09 | roswell | and it's even not friday's night, what's suprprising more... ehm |
| 18:19.37 | jmetro | my sip trunking stopped registering how does fix? |
| 18:20.03 | Katty | i hear chocolate does well when things aren't registering properly. |
| 18:20.33 | jmetro | I have a bag of 32 lindt chocolates. #jealousy |
| 18:20.34 | *** join/#asterisk _Corey_ (~chatzilla@63.133.202.2) |
| 18:20.43 | Katty | yum! |
| 18:20.54 | Katty | i have a bag of chocolate from sam's club, leftovers from the candybar cheesecake |
| 18:21.07 | *** join/#asterisk kresp0 (~kresp0@136.Red-95-120-50.dynamicIP.rima-tde.net) |
| 18:25.08 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 18:26.49 | Chainsaw | Katty: Excellent. You can never have too much chocolate :) |
| 18:27.28 | roswell | this choco story reminds me that from tbb... |
| 18:27.58 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 18:28.37 | roswell | "i had to take another shower... it wasn't enough... nothing will ever be enough" |
| 18:35.46 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 18:35.57 | jansiva | Hi |
| 18:37.28 | navaismo | wasaaaaaaaaaa |
| 18:38.07 | jansiva | navaismo: Hi |
| 18:38.56 | Penguin | navaismo: You are in Mexico? |
| 18:39.03 | navaismo | yes |
| 18:39.50 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 18:40.09 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 18:40.21 | jansiva | i am a new to asterisk |
| 18:40.32 | jansiva | what book you people recommend |
| 18:40.38 | Penguin | ~book |
| 18:40.38 | 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 |
| 18:40.45 | Penguin | That book ^^ |
| 18:40.45 | jansiva | we have lot of voip issue for the past 2 weeks |
| 18:41.13 | jansiva | i want a to figure out the issue |
| 18:42.50 | *** join/#asterisk eject_ck (~Evgeniy@109.86.157.219) |
| 18:43.14 | jansiva | we have lot of call choppy any good way to figure out |
| 18:43.29 | eject_ck | Hi all, I'm looking for Tollfree US forwarding to my SIP server, can I get recommendations there ? |
| 18:43.41 | eject_ck | ~ 3000 minutes per month |
| 18:44.32 | Penguin | jansiva: It's probably caused by jitter. |
| 18:45.51 | navaismo | jansiva, network congestion, did you have voice separated from data in your network, do you have QoS? |
| 18:45.59 | jansiva | Penguin: people are asking for proof for this |
| 18:46.31 | jansiva | the voip server is connect separately |
| 18:46.35 | jansiva | to the internet |
| 18:46.44 | Penguin | jansiva: Configure your PC to the voice vlan and check it with pingtest.net |
| 18:46.46 | jansiva | 5MB bandwidth |
| 18:48.24 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:48.26 | navaismo | how many calls? Which codec? |
| 18:49.06 | *** join/#asterisk Nix414s (~jubiejank@vpn.bctconsulting.com) |
| 18:50.45 | jansiva | is any way to monitor the end node grandstream phone using |
| 18:51.11 | Penguin | Does this mean listen to the phone call? |
| 18:53.15 | jansiva | G727 codec |
| 18:54.00 | jansiva | end use complaind his voice qulity is choppy at times when i say internet issue they need proof for this |
| 18:54.15 | Penguin | Does this mean you want to listen to the phone call? |
| 18:54.37 | *** join/#asterisk Nix414s (~jubiejank@vpn.bctconsulting.com) |
| 18:54.41 | jansiva | not listen |
| 18:54.48 | Penguin | Record to a file? |
| 18:55.06 | jansiva | i have to say this is the reason for voice choppy |
| 18:55.25 | jansiva | they expect proof |
| 18:55.38 | Penguin | How are you going to have the proof without hearing it? |
| 18:55.43 | Penguin | You can't see choppiness. |
| 18:56.19 | Katty | breaks out the alaskan knife |
| 18:56.27 | jansiva | then how will i prove the issue in internet |
| 18:56.31 | Katty | let me at it! we can see choppiness then! |
| 18:56.45 | Penguin | Have you already performed the pingtest at pingtest.net? |
| 18:56.58 | navaismo | tcpdump, mtr |
| 18:58.05 | Katty | http://objectplanet.com/probe/ <- or that. |
| 18:58.37 | Katty | or i guess you could use Appnetta. |
| 18:58.49 | Katty | nChronos, Capsa, Solarwinds |
| 18:58.58 | Katty | cyberoam firewalls have network monitor |
| 18:59.01 | Katty | sophos probably does too |
| 18:59.13 | *** join/#asterisk kresp0 (~kresp0@136.Red-95-120-50.dynamicIP.rima-tde.net) |
| 19:07.37 | qakhan | new issue :P |
| 19:07.58 | qakhan | is there any way to remove beep on call waiting ? |
| 19:08.47 | navaismo | i think that is a phone side configuration |
| 19:10.16 | qakhan | yes we are using Polycom 670 |
| 19:11.37 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 19:14.26 | [TK]D-Fender | qakhan: You can disable it in provisioning |
| 19:14.28 | jansiva | is any opensource tool to monitor choppy ness |
| 19:15.38 | navaismo | tcpdump, mtr |
| 19:17.55 | Penguin | jansiva: Why have you ignored the answers you've been given? |
| 19:19.53 | jansiva | I have not ignored anything |
| 19:20.01 | Penguin | (1356.45) <Penguin> Have you already performed the pingtest at pingtest.net? |
| 19:20.06 | Penguin | No response from you. |
| 19:20.14 | Penguin | (1356.58) <navaismo> tcpdump, mtr |
| 19:20.19 | Penguin | No response from you on that, either. |
| 19:20.52 | Penguin | The test at pingtest.net will show jitter. |
| 19:21.15 | jansiva | got you |
| 19:21.51 | Penguin | mtr will show you min and mix pings, and you can determine the jitter in that manner as well. |
| 19:21.55 | *** join/#asterisk jsjc (~Adium@119.Red-83-39-26.dynamicIP.rima-tde.net) |
| 19:22.18 | jansiva | make sense |
| 19:23.40 | Penguin | I like mtr a lot because it can use udp mode and I can follow the path to my ITSP. |
| 19:25.03 | *** join/#asterisk imcdona (~Thunderbi@gateway.defiancedc.com) |
| 19:25.19 | jansiva | i have a question my server could be located in one location and when i do pingtest.net, this server can be located in other location |
| 19:25.44 | Penguin | If I watch the "Last" column, I can see how much the ping time fluctuates. |
| 19:25.54 | jansiva | then how do we analyses this |
| 19:26.15 | Penguin | Do you have the hostname for the provider? mtr -tu sip.yourprovider.net |
| 19:26.17 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:26.17 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:26.23 | Penguin | Watch the ping times. |
| 19:27.12 | Penguin | Mine is +/- 3 |
| 19:31.21 | jansiva | what is +/- 3 |
| 19:34.10 | *** join/#asterisk Nix414s (~jubiejank@vpn.bctconsulting.com) |
| 19:34.15 | Penguin | If I use udp datagrams to test latency between my asterisk computer and my ITSP, the ping times are on average 21.2ms. But sometimes they are as low as ~19 and sometimes as high as ~24. |
| 19:34.16 | *** part/#asterisk Nix414s (~jubiejank@vpn.bctconsulting.com) |
| 19:34.24 | Penguin | So I have 6ms jitter. |
| 19:34.46 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 19:35.47 | Penguin | I saw one ping that went up to 28.9, so I have 10ms jitter. |
| 19:36.17 | Penguin | But 10ms jitter is not going to give me quality issues. |
| 19:38.29 | *** join/#asterisk boch (~boch@186.182.116.17) |
| 19:39.48 | boch | hi all.. quick question, how can i trace whena enters or leaves music on hold? |
| 19:40.12 | boch | <PROTECTED> |
| 19:40.57 | Penguin | If you have too much jitter, a jitter buffer may be able to help you. |
| 19:41.09 | Penguin | boch: core set verbose 3 |
| 19:41.50 | anonymouz666 | If you have too much jitter buffer, delay may be able to reach you. |
| 19:42.52 | Penguin | And if you have too much delay, it's hard to carry on a conversation. |
| 19:43.19 | boch | Penguin, there is no queue_log entry or something like that ? |
| 19:43.27 | jansiva | sorry |
| 19:43.53 | jansiva | for me to understand and analyzing taking some time |
| 19:44.24 | jansiva | even when i say internet issue people are not believing me |
| 19:44.33 | Penguin | http://en.wikipedia.org/wiki/Packet_delay_variation |
| 19:44.39 | jansiva | i mean my management they say it is network issue |
| 19:45.14 | Penguin | It _IS_ a network issue. |
| 19:46.58 | *** join/#asterisk navaismo (~navai_000@189.191.249.45) |
| 19:47.08 | Penguin | Jitter is caused by network congestion. |
| 19:48.15 | jansiva | where do i see in mtr |
| 19:48.32 | jansiva | i see Loss% Snt Last Avg Best Wrst StDev |
| 19:48.38 | jansiva | these colums |
| 19:48.40 | anonymouz666 | mtr uses ICMP, right? |
| 19:48.44 | Penguin | no |
| 19:48.49 | Penguin | it CAN use icmp |
| 19:48.49 | anonymouz666 | UDP? |
| 19:49.12 | Penguin | mtr -u <---- use UDP datagrams instead of ICMP echo |
| 19:49.29 | anonymouz666 | ok, but what about the default mode? ICMP? |
| 19:49.33 | Penguin | yes |
| 19:49.55 | Penguin | jansiva: (1426.15) <Penguin> Do you have the hostname for the provider? mtr -tu sip.yourprovider.net |
| 19:50.13 | Penguin | Look at the Avg. Look at the Last. |
| 19:50.42 | Penguin | Watch how much the Last number fluctuates. |
| 19:51.20 | Penguin | If you see 19, 20, 19, 21, 22, 19, 22... you have about 3ms jitter. |
| 19:51.30 | anonymouz666 | Just to remember... some people use ICMP mode to measure the network. In fact, you can have an idea about the quality of network, but some routers, in the path, could treat UDP packets differently. Even the discard rule could consider UDP first. |
| 19:52.08 | anonymouz666 | Penguin: Until 20ms of jitter I think you are fine |
| 19:52.26 | Penguin | I'm seeing max 10ms on mine. |
| 19:52.42 | Penguin | Usually 5-6ms. |
| 19:59.35 | Penguin | The maximum variation I have had over the past 30 minutes of testing was 23ms. That's just Best vs. Wosrt. |
| 20:00.08 | Penguin | But there is one thing I'd like to ask about... |
| 20:00.42 | Penguin | If a router in the path has a much higher variation between best and worst, does that affect the call just as much as the jitter on the last node? |
| 20:01.42 | Penguin | There's one router right before I reach the ITSP has showed a 17.5 for best and a 150.1 for worst. On average it is only 20.9, though, and current jitter is only about 6ms. |
| 20:04.06 | jansiva | <PROTECTED> |
| 20:04.22 | jansiva | 4. xe-11-1-1.lon21.ip4.tinet.net 1.4% 73 152.3 164.4 152.2 330.2 31.3 |
| 20:04.22 | jansiva | <PROTECTED> |
| 20:04.42 | Penguin | You're showing a lot of packet loss. |
| 20:04.59 | Penguin | I show 0% packet loss during my testing over 30 minutes. |
| 20:05.28 | jansiva | when i address with my isp he say reverse routting issue |
| 20:08.54 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 20:11.34 | jansiva | when i ping the ip i dont see any packet loss |
| 20:11.44 | jansiva | when i put mtr i see packet loss |
| 20:19.43 | jmetro | some hops of the journey might not respond to icmp |
| 20:20.12 | jmetro | my default gateway wont respond to ping but i still get internet access |
| 20:20.29 | Penguin | But that won't show 15% packet loss. |
| 20:20.34 | jmetro | it would 100% |
| 20:20.43 | Penguin | It'll either show all of it or will show nothing at all there. |
| 20:22.19 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 20:39.37 | *** join/#asterisk Dovid (~Dovid@63.133.202.2) |
| 20:40.47 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 20:42.10 | *** join/#asterisk diatonic (~diatonic@unaffiliated/diatonic) |
| 20:44.46 | diatonic | So I logeed an issue last Friday. https://issues.asterisk.org/jira/browse/ASTERISK-22661 How long do these usually sit in triage? |
| 20:44.55 | jmetro | 525600 minutes |
| 20:45.20 | diatonic | awesome |
| 20:45.53 | pabelanger | diatonic: if it is important for you, you'll likely have to hire a developer to fix it |
| 20:45.56 | pabelanger | otherwise |
| 20:46.11 | pabelanger | ~patience |
| 20:46.11 | infobot | i guess patience is a Godly attribute, or the solution for most things. |
| 20:46.30 | volga629 | I have some strange issue I can't get TLS client connect to asterisk without pedantic=no |
| 20:47.17 | diatonic | I was just curious how long digium typically takes to classify or verify or something :/ |
| 20:47.33 | volga629 | without this setting it just dropping registration each 10 sec |
| 20:47.48 | pabelanger | diatonic: they have development sprints every month |
| 20:47.56 | pabelanger | but there is over 900 issues in front of you |
| 20:48.05 | pabelanger | so, you can infer from that |
| 20:48.13 | diatonic | thanks |
| 20:52.49 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
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| 21:09.42 | *** join/#asterisk crosstalk (~adnan@chello062178127150.4.13.vie.surfer.at) |
| 21:09.47 | crosstalk | Hi All |
| 21:10.26 | crosstalk | can anyone please help me on a damn bad IVR Crosstalk problem |
| 21:10.28 | crosstalk | :( |
| 21:11.44 | crosstalk | I googled it for two days |
| 21:11.48 | crosstalk | but no luck |
| 21:13.13 | ChannelZ-Wk | Crosstalk? |
| 21:16.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:16.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:18.20 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:22.06 | *** join/#asterisk crosstalk (~adnan@chello062178127150.4.13.vie.surfer.at) |
| 21:22.22 | crosstalk | Sorry ChannelZ-Wk |
| 21:22.35 | crosstalk | my laptop was not responding |
| 21:22.51 | crosstalk | not crosstalk from other channel |
| 21:23.01 | crosstalk | we have a calling card platform |
| 21:23.08 | crosstalk | so when customer calls a DID |
| 21:23.27 | crosstalk | an IVR prompts like: Please enter your PINcode bla bla |
| 21:23.52 | crosstalk | but sudeenly there comes a Male IVR and starts interrupting the whole IVR process |
| 21:24.19 | crosstalk | and if the user sends DTMF during this occurs , the DTMFS are not being read correctly |
| 21:24.29 | crosstalk | it happens once in a 3 calls |
| 21:25.19 | crosstalk | so the callers are getting really irritated as they have to call again and start the whole IVR process |
| 21:25.44 | crosstalk | any1, any idea will be appreciated more than anything |
| 21:26.12 | ChannelZ-Wk | And what is this 'male IVR' saying? It's not something from your system? |
| 21:26.25 | crosstalk | the male IVR keeps saying |
| 21:26.32 | crosstalk | Please enter your PIN Code |
| 21:26.37 | crosstalk | your PIn is invalid |
| 21:26.43 | crosstalk | please enter your PIN code |
| 21:26.45 | ChannelZ-Wk | But it's not your prompt? |
| 21:26.48 | crosstalk | nope |
| 21:26.55 | crosstalk | we never set this male voice |
| 21:27.06 | ChannelZ-Wk | Well it has to be coming from somewhere, maybe your telco is screwing something up |
| 21:27.17 | crosstalk | we set a female voice which actually should prompt this |
| 21:27.36 | ChannelZ-Wk | You say "we set a female voice" -- is this text-to-speech or audio recordings? |
| 21:27.36 | crosstalk | well you mean telco from where the DID is being served |
| 21:27.49 | crosstalk | call it starts before we terminate the call |
| 21:28.01 | crosstalk | its an audio recording |
| 21:28.30 | crosstalk | default asterisk prompt i think |
| 21:28.47 | ChannelZ-Wk | OK well Asterisk doesn't have the ability to just make men start speaking on your calls so you've got to figure out where it's coming from. |
| 21:29.05 | crosstalk | i turned the full logging on |
| 21:29.08 | ChannelZ-Wk | What is the path of one of these calls, from the client to asterisk |
| 21:29.10 | crosstalk | but trust me no trace |
| 21:29.24 | crosstalk | ok let me tell you |
| 21:29.27 | crosstalk | the path |
| 21:29.33 | ChannelZ-Wk | I trust you because it's not something Asterisk "just does" |
| 21:30.21 | crosstalk | client calls the DID (virtual)--> DID Provider sends it to our SIP PROXY (Opensips)--> OpenSips balances the load between two asterisks |
| 21:30.45 | crosstalk | and asterisk takes over the media stream |
| 21:30.57 | crosstalk | but before we start leg B it all ahppens |
| 21:31.15 | crosstalk | so it should have nothing to do with the careers that terminate our calls |
| 21:31.38 | crosstalk | it is between the DID provider and our Asterisk Servers |
| 21:31.49 | ChannelZ-Wk | well I'm guessing it has to be happening at the telco side if anything |
| 21:32.27 | crosstalk | how can I trace that |
| 21:32.33 | crosstalk | any idea? |
| 21:32.52 | crosstalk | one cool idea can be taking DID from other provider and test |
| 21:32.56 | crosstalk | what do you think |
| 21:33.12 | ChannelZ-Wk | ?? Make a bunch of test calls until you can get it to happen and then get on the line with them so they can look |
| 21:33.34 | ChannelZ-Wk | Record the call, send them a sample |
| 21:33.43 | ChannelZ-Wk | It seems bizarre |
| 21:34.08 | crosstalk | i just send them the recording |
| 21:34.13 | crosstalk | they returned me saying |
| 21:34.24 | ChannelZ-Wk | and oddly specific that this male voice is asking for PINs |
| 21:34.27 | crosstalk | well this has to be something to be done with your IVR system |
| 21:34.32 | crosstalk | yes |
| 21:35.09 | crosstalk | well i guess asking for PIn is a rendom phenomenon I think |
| 21:35.14 | ChannelZ-Wk | This audio is mixed with what should be the real call audio, or is it chopy or screwed up or anything? |
| 21:35.30 | crosstalk | totally mixed |
| 21:35.40 | crosstalk | like 3 sec male |
| 21:35.46 | crosstalk | then again steffany |
| 21:35.49 | crosstalk | and so on |
| 21:37.08 | crosstalk | the DID provider gave me few lessons saying: you need to talk to the provider of your IVR etc etc |
| 21:37.20 | ChannelZ-Wk | in order for asterisk to inject some other audio into a call like that, something very specific has to occur... this is digital. You're not using ChanSpy anywhere, or shared call appearances, or anything like that? |
| 21:37.20 | crosstalk | this system is in Production |
| 21:37.56 | crosstalk | shared call no. |
| 21:38.04 | crosstalk | chanspy I am not sure |
| 21:38.18 | crosstalk | can it have something to do with chanspy |
| 21:38.20 | crosstalk | ? |
| 21:39.15 | crosstalk | do you mean shared call whether the DID is being shared, right? |
| 21:39.24 | crosstalk | if yes NO it is dedicated to us |
| 21:43.31 | ChannelZ-Wk | well Chanspy is one way that audio can get mixed into a stream from some other source |
| 21:43.47 | crosstalk | ok very imp option |
| 21:43.55 | crosstalk | let me deactivate it globally |
| 21:43.59 | crosstalk | on both servers |
| 21:44.13 | crosstalk | are you always here ChannelZ-Wk ? |
| 21:44.32 | crosstalk | so that i can tell you the results |
| 21:44.33 | crosstalk | :) |
| 21:44.38 | ChannelZ-Wk | and shared calls as in, say you have 3 "lines" and you have 5 phones with line keys, you can set it up such that pressing line key 1 on any phone can pick up that 'line' |
| 21:44.51 | crosstalk | understood |
| 21:45.05 | ChannelZ-Wk | It's accomplished by putting the call into a conference of sorts that any of the phones can then join. So it's another way for audio to get into a call from somewhere else. |
| 21:45.35 | crosstalk | but i suspect its gotta be chanspy |
| 21:45.37 | ChannelZ-Wk | otherwise I can't imagine what could be going on local to your system |
| 21:45.54 | crosstalk | i deactivated it, and made 5 calls while talking to you |
| 21:45.58 | crosstalk | and no amle IVR yet |
| 21:46.09 | ChannelZ-Wk | So how are you calling ChanSpy, and why? |
| 21:46.33 | crosstalk | i suspect a coleague was developing a monitoring addon |
| 21:46.36 | ChannelZ-Wk | IE are you using it just to monitor calls, or what arguments are you passing ChanSpy |
| 21:46.58 | crosstalk | i have no idea honestly, he is doing this |
| 21:47.05 | crosstalk | but what i could i have done |
| 21:47.11 | ChannelZ-Wk | (and yes I'm usually here, as ChannelZ and/or ChannelZ-Wk) |
| 21:47.23 | crosstalk | deactivated chanspy on both systems |
| 21:47.34 | crosstalk | thanks in advance |
| 21:47.53 | crosstalk | nope |
| 21:47.57 | crosstalk | it happened again |
| 21:48.00 | crosstalk | right now |
| 21:48.01 | crosstalk | .( |
| 21:48.11 | crosstalk | grrr |
| 21:48.38 | ChannelZ-Wk | do you have verbose console turned on? |
| 21:49.00 | ChannelZ-Wk | Make sure something weird in your dialplan isn't going on (sounds like you have multiple people with access) |
| 21:49.48 | crosstalk | on these asterisk system no UAs are connected |
| 21:50.00 | crosstalk | only calling card dialplans |
| 21:52.26 | crosstalk | yeah console is on verbose, |
| 21:53.01 | crosstalk | console => notice,warning,error |
| 21:53.01 | crosstalk | messages => notice, warning, error |
| 21:53.01 | crosstalk | full => notice, warning, error, debug, dtmf |
| 21:53.13 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 22:04.07 | *** join/#asterisk xzarth (~krikkit@dh207-15-243.xnet.hr) |
| 22:19.24 | *** join/#asterisk kemmler (tully@r74-192-178-52.htvlcmta01.hnvitx.tl.dh.suddenlink.net) |
| 22:21.06 | kemmler | Do you have to use the default context if you call a local channel? |
| 22:21.19 | kemmler | from a .call file or originate |
| 22:23.15 | kemmler | eg "Channel: Local/66\nContext: testcall\nExtension: 67 |
| 22:26.51 | ChannelZ-Wk | isn't there an @somewhere syntax for Local? can't remember I don't use locals |
| 22:27.19 | WIMPy | local/extension@context |
| 22:27.36 | WIMPy | And good morning, BTW. |
| 22:31.12 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:31.42 | crosstalk | good morning |
| 22:37.00 | crosstalk | for local UAS |
| 22:37.03 | crosstalk | i have |
| 22:37.08 | crosstalk | [mor_local] |
| 22:37.08 | crosstalk | exten => _+.,1,Goto(${EXTEN:1},1) |
| 22:37.08 | crosstalk | exten => fax, 1, Goto(mor_fax2email,${EXTEN},1) |
| 22:37.08 | crosstalk | switch => Realtime/mor_local@extensions |
| 22:37.32 | kemmler | Channel: Local/66@testcall\nContext: testcall\nExtension: 67 |
| 22:37.37 | kemmler | throws this error No such extension/context 66@default while calling Local channel |
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| 23:10.10 | Penguin | kemmler: Make sure you have extension 66 in default context. |
| 23:10.39 | kemmler | Penguin, yeah i gather that. I was wondering if there was a way to have it in something other than the default context |
| 23:11.02 | Penguin | Of course. Define the context to use. |
| 23:11.18 | kemmler | <kemmler> Channel: Local/66@testcall\nContext: testcall\nExtension: 67 |
| 23:11.18 | kemmler | <kemmler> throws this error No such extension/context 66@default while calling Local channel |
| 23:11.33 | kemmler | i defined it there and it still threw the error |
| 23:11.57 | Penguin | I see what you're saying. |
| 23:12.48 | kemmler | on 1.6 btw |
| 23:13.07 | Penguin | That part isn't helpful. |
| 23:13.12 | WIMPy | Try #history |
| 23:13.27 | Penguin | And regardless of what branch it is, it isn't supported and there have been many bugs fixed since then. |
| 23:14.10 | kemmler | work machine, i'm supporting it or trying to anyway |
| 23:15.25 | Penguin | Pastebin your entire call file and I'll throw it into my 1.8 box and see if it behaves correctly. |
| 23:15.44 | kemmler | kk sec |
| 23:19.05 | Katty | hi kids. |
| 23:19.18 | WIMPy | Hi mum! |
| 23:19.28 | Katty | how're you dear |
| 23:19.38 | WIMPy | When's dinner ready? |
| 23:19.56 | WIMPy | Well, actually I had too much of that already. |
| 23:20.02 | Katty | mmm, bout 15 minutes i think |
| 23:20.11 | Katty | what did you have? |
| 23:20.52 | WIMPy | Made a little mistake in ordering the house salad at our developers regulars table after I had dinner at my other mum. |
| 23:21.22 | WIMPy | Unfortunatly the salad is more to eat than the usual meaty stuff. |
| 23:21.30 | kemmler | Penguin, http://pastebin.com/FwQ8jbck |
| 23:21.55 | Penguin | Channel: Local/70 |
| 23:21.59 | Penguin | This means default context. |
| 23:22.04 | WIMPy | kemmler: I don't see a context in there. |
| 23:22.36 | Penguin | If you want something else, you have to specify Local/70@somethingnotdefault |
| 23:23.07 | Katty | ah, right. |
| 23:23.16 | Katty | well, at least you didn't go for the potato skins or something |
| 23:23.24 | Penguin | Gah! I will never buy another Yaesu radio. |
| 23:23.25 | Katty | you'd /really/ be stuffed |
| 23:23.40 | WIMPy | I am |
| 23:23.50 | WIMPy | But I'm not much in to potatoes. |
| 23:24.13 | Penguin | I like them baked next to my steak. |
| 23:27.27 | Penguin | kemmler: If I define the context on the local channel, it works correctly for me. If I do not define it, such as you didn't, it behaves exactly as I said. |
| 23:28.04 | kemmler | You're right Penguin I just got it working, sorry |
| 23:28.15 | Penguin | No problem. |
| 23:28.41 | Katty | WIMPy: ah right. |
| 23:28.46 | Katty | WIMPy: also...you're odd hehe |
| 23:28.56 | Katty | WIMPy: but hey, more potatoes for me i suppose! woo! |
| 23:29.53 | WIMPy | You're welcome. |
| 23:37.13 | kemmler | is the default agi-bin directory past 1.6 /usr/share/asterisk/agi-bin ? |
| 23:37.46 | Penguin | It's whatever you set in asterisk.conf. |
| 23:38.07 | Penguin | /var/lib/asterisk/agi-bin is the typical dir. |
| 23:38.13 | kemmler | ahh gotcha |
| 23:38.19 | kemmler | thanks again |
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