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| 00:39.03 | Atom0 | my server doesn't seem to support the "100 trying" response that my SIP is giving and immediately sends out a CANCEL request. is there any file I can edit to add the code 100? |
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| 00:40.59 | Atom0 | "SIP/2.0 100 trying -- your call is important to us" |
| 00:41.20 | Atom0 | as soon as it receives that my server cancels the call within milliseconds. |
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| 00:47.07 | jdzielny | hello everyone. Does anyone have any ideas how one might go about changing the voicemail message filenaming convention so that instead of msgXXXX.wav, it would be something like the call monitor function is? |
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| 00:58.59 | asfp | If I want a dial plan triggered by an XMPP message being sent to Asterisk, how do I do it? I see that there is specific support for gtalk, but I only want to start an action. |
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| 02:32.43 | jdzielny | hello everyone. Does anyone have any ideas how one might go about changing the voicemail message filenaming convention so that instead of msgXXXX.wav, it would be something like the call monitor function is? |
| 02:33.24 | WIMPy | Change the source. |
| 02:35.04 | WIMPy | And I predict a hard time if you try to put anything non-static in front of the XXXX. |
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| 02:54.13 | mjordan | I predict a hard time if you change the XXXX |
| 02:54.28 | mjordan | really, it's just a whole lot of "you're going to have a bad time" |
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| 03:34.58 | jdzielny | "If you think you can recode Asterisk, you're going to have a bad time" |
| 03:35.00 | jdzielny | lol |
| 03:39.05 | jdzielny | Maybe one way to do it without recoding the source would be to write a script that creates hardlinks with filenames based on the metadata in the msgXXXX.txt files |
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| 03:58.06 | jdzielny | wb mjordan |
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| 08:11.27 | PaybackTony | Anyone in here ever see / get past an issue with compiling asterisk (11.11 in my case) with ICE support? I installed uuid-dev and libuuid1, but still not getting ICE offers (as needed for WebRTC / Sip.js). |
| 08:13.05 | PaybackTony | On Ubuntu 14 that is |
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| 08:27.36 | PaybackTony | Nevermind |
| 08:27.52 | PaybackTony | I compiled latest asterisk (11.13) LTS and ICE support works |
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| 09:00.08 | cmendes0101 | If I have the linkedid, is there an easy way to get the current channel name? |
| 09:03.53 | pukkita_ | hi |
| 09:06.02 | pukkita_ | running an outgoing campaign, looking for some way to show the callerid of the called number when the dialer returns it to the outbound queue |
| 09:06.43 | pukkita_ | problem is I don't know if that would be stored on any variable |
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| 09:14.04 | Stefan27 | [2014-11-05 10:00:11] WARNING[22790]: asterisk.c:3698 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible! |
| 09:14.04 | Stefan27 | ! THIS is an EX-CANARY. (Reducing priority) |
| 09:14.07 | ChannelZ | The callerid of the _called_ number? That's not something that is normally know |
| 09:14.08 | Stefan27 | What's this?? |
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| 09:14.21 | Stefan27 | What's the purpose of canary in asterisk? |
| 09:14.36 | ChannelZ | The canary is kind of a heartbeat |
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| 09:15.40 | ChannelZ | It monitors asterisk for threads that go out of control |
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| 09:16.12 | ChannelZ | But you must have been running asterisk in realtime priority for that to occur |
| 09:49.57 | Stefan27 | I did run it in rt |
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| 09:51.29 | ChannelZ | Well then it means something started sucking CPU and starved everything else out for a period of time, causing the canary to die. |
| 09:51.45 | Stefan27 | Does anyone have any experience of starting/stopping/monitoring asterisk with systemd asterisk.service file? I'm using fedora 20 and I don't have that /etc/init.d/asterisk script for starting and stopping asterisk anymore... I use ExecStart=/usr/sbin/safe_asterisk -U asterisk -G asterisk and ExecStop=/usr/bin/asterisk -rx 'core stop now' but the latter is not sure to kill asterisk |
| 09:52.34 | Stefan27 | I guess I need to create a script that does 'core stop now' first and if that fails wait some time and kill it harder - there must be such scripts out there already? |
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| 09:53.06 | jamesc | in the astierks source directory where does the init.d scripts live? |
| 09:53.15 | Stefan27 | LOL |
| 09:53.20 | Stefan27 | i was about to ask that james |
| 09:53.44 | jamesc | cool that make two of us :) |
| 09:54.38 | Stefan27 | maybe i can use the init.d/asterisk script for fedora20? wrapping it into systemctl service somehow |
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| 09:56.00 | jamesc | Stefan27: roll on systemd hopefully will put an end to all this init shit |
| 09:56.27 | Milos | ChannelZ, did you see my message from a few days ago |
| 09:56.35 | jamesc | no? |
| 09:56.59 | jamesc | Stefan27: I found them there in contrib./init.d |
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| 10:34.38 | pukkita_ | is there any command from the CLI to remove sip sessions? I see lots of stale ones |
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| 10:50.36 | LooserOuting | Hi. when you run tests/channels/SIP/directrtpsetup from asterisk test suite at one point phone A will hang up but asterisk sends INVITES to phone B instead of a bye. How can I turn on debugging when starting the test ? |
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| 11:21.57 | marceloamorim | there any tip to get faster detect from leg B when I do a call from my asterisk using dahdi and digium wildcard TDM410P |
| 11:22.21 | WIMPy | Scrap it and go digital. |
| 11:23.00 | marceloamorim | I`m from Brazil and almost all lines are analogs =( |
| 11:23.35 | WIMPy | There's always the SIP shit. |
| 11:23.54 | marceloamorim | I`m getting the audio when the IVR are like number 7 |
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| 12:01.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 12:39.37 | marceloamorim | I don`t know if it is a sip problem once I put 'r' option on app dial, and keep ringing until IVR are at number 7 |
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| 12:51.47 | olinux | updated to switchvox 5.9 from 5.8.5, was fine last week, now rebooting or software restarting randomlyprobably 6 times in past 6 hours |
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| 14:46.07 | pukkita_ | hi, I'm trying to track down the called number on an outgoing campaign. A predictive dialer do the ougoing calls, then when one is answered passes it to an outgoing queue that in turn calls the (dynamic) agents. Problem is agents only have (hard)phones (no PC) and the callerid that the agents see is our outgoing CID, so I'm looking to modify the dialplan so that it swaps the CID with the proper (dialled) one. |
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| 14:48.05 | [TK]D-Fender | So go change the callerid in your dialplan before it gets sent to the queue. |
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| 15:29.05 | anonymouz666 | strange... i have a queue configured with 20 secs timeout. when it reachs the 20 secs, the abandon event comes with 15 secs value. |
| 15:29.40 | anonymouz666 | timeout in this case should be equal to abandon secs |
| 15:29.52 | anonymouz666 | am I missing something here? |
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| 15:52.07 | anonymouz666 | got disconnected |
| 15:56.42 | pukkita_ | <[TK]D-Fender>: Problem I'm facing is as the call progress in the dial plan and jumps from context to context I'm not being able to track it, Tried vars with __ to no avail, now trying using DB(), but the problem is what could I use as key to "track" the call |
| 15:59.45 | pukkita_ | need something to track the call from context to context |
| 16:00.11 | pukkita_ | to set the callerid prior to hand the call to the queue |
| 16:02.54 | [TK]D-Fender | There is no such thing as tracking context to context |
| 16:03.04 | [TK]D-Fender | Channel vars are always the same. |
| 16:03.11 | [TK]D-Fender | There is no concept of "scope" |
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| 16:10.32 | pukkita_ | what I mean is the dialer calls out, source channel being SIP/EXT-xxxxxx, in that context I can set the called number to a variable or a DB. Once is answered, a new channel is created (SIP/trunk-xxxxx) then channels are bridged |
| 16:13.03 | [TK]D-Fender | If it calls SIP/device to call your device first, then proceeds to place the outbound call then you are going to either have to set the callerID in whatever you use to originate the call (, or use CONNECTEDLINE to update the callerid. |
| 16:14.02 | pukkita_ | but don't know what property or variable could I use as a key for so that I'm able to set the callerID just before hading the call to the queue or agent |
| 16:14.37 | pukkita_ | the callerid is set to our main number |
| 16:15.04 | [TK]D-Fender | you set the callerID in whatever you are originating from, or set it useing CONNECTEDLINE in the dialplan they gt dumped into. |
| 16:15.06 | pukkita_ | system dials out (dialer) and sets the callerid properly to our main number |
| 16:15.59 | pukkita_ | connectedline? gonna check |
| 16:16.17 | pukkita_ | thanks! |
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| 16:18.12 | ChannelZ | Milos: Yes - something about something fixing something but I didn't remember what problem you had/were reffering to. |
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| 16:18.27 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-crwkshstzxwkbgab) |
| 16:18.29 | Milos | ChannelZ, on android 4.4.4 and earlier the tag header is missing from CANCEL. |
| 16:18.38 | ChannelZ | oh, that. |
| 16:18.42 | Milos | pedantic=no makes this work because pedantic=no tells the server not to validate tags |
| 16:18.43 | Milos | :D |
| 16:18.50 | Milos | yay workarounds... |
| 16:19.47 | ChannelZ | Lovely. Terrible name but whatever works |
| 16:19.53 | Milos | indeed |
| 16:31.46 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:221:6aff:feb8:e0b2) |
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| 16:56.41 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:57.29 | net_ | is there any popup application where i can display a variable in that popup when there is a incoming call? |
| 16:57.46 | net_ | apart from vicidial or vtiger or sugarforge.... |
| 16:58.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-sabvwjdfsnqsvefw) |
| 16:58.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:25.25 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-94-139.user.veloxzone.com.br) |
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| 17:36.51 | *** join/#asterisk lillian (~lillian@unaffiliated/lillian) |
| 17:37.49 | lillian | Using Polycom phones. We all get the problem whereas if you ATTEMPT to transfer someone to an extension's voicemail, e.g., hitting transfer button then entering "*1010" it'll ask you to "enter more digits" on the polycom phone screen, and if I hit dial or whatever it brings ME to the voicemail |
| 17:38.18 | lillian | it's the only problem we've had |
| 17:41.35 | [TK]D-Fender | Fix your dialplan on the phone |
| 17:42.37 | lillian | Oh dear, any tips? |
| 17:42.44 | lillian | i was afraid it'd be complicated |
| 17:43.04 | lillian | because we can transfer to extensions just fine, and dial into voicemail fine, just not... transfer to voicemail? |
| 17:44.50 | [TK]D-Fender | There is no such thing as "transfer to voicemail" |
| 17:44.54 | [TK]D-Fender | You transfer to an EXTENSION. |
| 17:45.00 | [TK]D-Fender | One is no different from the other |
| 17:45.04 | [TK]D-Fender | Make it match properly |
| 17:45.18 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:47.36 | *** join/#asterisk n3tctrl (~n3tctrl@12.218.71.199) |
| 17:49.34 | n3tctrl | has anybody ever connected an asterisk based server to an overhead paging system using FXO trunk port? |
| 17:50.19 | [TK]D-Fender | Lots of people have. |
| 17:50.33 | WIMPy | n3tctrl: I'm sure someone has tried that at some point. |
| 17:50.56 | n3tctrl | okay. just wanted to make sure it would actually work. we have a paging system that is connected to a 30 year old pbx they want me to replace. |
| 17:52.26 | [TK]D-Fender | You just stated it was an FXO |
| 17:52.32 | [TK]D-Fender | * has many different FXO interfaces |
| 17:52.44 | [TK]D-Fender | FXO is FXO.. therefor * can use your paging system |
| 17:52.47 | n3tctrl | using grandstream gxw4108 |
| 17:54.18 | [TK]D-Fender | * can clearly talk to that device... |
| 17:55.05 | n3tctrl | cool! they wanted me to use a shoretel setup, but i found asterisk, and even going with one of the commercial asterisk based solutions, it is far cheaper! |
| 17:59.03 | babak | Hi,has anybody worked with DAHDIRAS ? |
| 17:59.47 | WIMPy | Again... |
| 17:59.52 | WIMPy | ~polls |
| 17:59.52 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 18:01.26 | babak | is DAHDIRAS working in new versions of Asterisk ? we should run DAHDRAS both in server and client system ? |
| 18:05.02 | *** join/#asterisk Scrix0 (~Scrix0@rrcs-208-125-125-114.nys.biz.rr.com) |
| 18:05.05 | [TK]D-Fender | babak: If it's there then it should be assumed to be working. |
| 18:05.20 | [TK]D-Fender | babak: No-one is deliberately including modules that are known not to work. |
| 18:06.14 | Scrix0 | Hello - I'm very new to Asterisk. We're trying to implement Asterisk RT however we're running into some intermittent issues. Over a period of time, calls sent from one PBX to another instantly go to voicemail |
| 18:06.45 | babak | [TK]D-Fender: wish to find a small guide on official wiki or mailing list .. |
| 18:07.43 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DAHDIRAS |
| 18:08.04 | file | I don't think anyone has touched/tried DAHDIRAS in years |
| 18:08.35 | WIMPy | hasn't tried anything dialup since I4L. |
| 18:10.08 | babak | file: not bad idea you can connect and manage your Asterisk servers remotely via E1(PRI) links when data link is not available |
| 18:10.57 | WIMPy | I've got some hardware NAS, if you need any. |
| 18:13.17 | Scrix0 | Does anyone know if rtautoclear necessary if rtcachefriends is enabled? |
| 18:41.54 | lillian | [TK]D-Fender: kk |
| 18:44.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:55.23 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 18:55.43 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 18:55.48 | qakhan | hi all, is there any way we can detect voice mail on outbound call? |
| 18:56.09 | [TK]D-Fender | "core show application AMD" |
| 18:56.11 | [TK]D-Fender | ^ |
| 18:57.04 | qakhan | do i use AMD() after Dial() or before it? |
| 18:57.18 | [TK]D-Fender | What do you think? |
| 18:57.29 | qakhan | after Dial() |
| 18:57.54 | [TK]D-Fender | if you want to do this for the CALLEE.. then doing it before the Dial is pointless |
| 19:01.19 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xvxvytetmyrowwyc) |
| 19:01.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:01.41 | anonymouz666 | qakhan: After Dial and works only if the channel answers it. |
| 19:02.30 | anonymouz666 | what happens in pre-dial state is ignored |
| 19:02.40 | *** join/#asterisk benasse (~benasse@cicogna.fr) |
| 19:03.24 | *** join/#asterisk wizhippo (~wizhippo@64.201.57.7) |
| 19:04.39 | [TK]D-Fender | Not "ignored". Perhaps not relevant.... |
| 19:05.22 | anonymouz666 | It IS relevant. Could exist a lot of information before Answer signalling. |
| 19:05.52 | anonymouz666 | if you do not consider that, good chance you will deliver a lot of voicemail boxes to "your" operators |
| 19:08.07 | [TK]D-Fender | anonymouz666: That'd be the "relevent" part I was talking about. |
| 19:08.27 | [TK]D-Fender | anonymouz666: Doing it to the CALLER doesn't mean it would be ignored... just completely not what he wants :) |
| 19:08.35 | [TK]D-Fender | anonymouz666: Details are in the big print :) |
| 19:12.28 | qakhan | [TK]D-Fender here is my config |
| 19:12.29 | qakhan | http://pastebin.com/aqn1wt7S |
| 19:12.49 | [TK]D-Fender | [13:57][TK]D-FenderWhat do you think? |
| 19:12.51 | [TK]D-Fender | [13:57]qakhanafter Dial() |
| 19:13.00 | qakhan | it shows machine |
| 19:13.03 | [TK]D-Fender | You just said "after"... so why are we seeing you doing it BEFORE? |
| 19:13.51 | qakhan | it does not get on AMD() after Dial() and call hangup |
| 19:15.47 | [TK]D-Fender | qakhan: So what good do you think putting it BEFORE does? |
| 19:16.01 | [TK]D-Fender | qakhan: You don't seem to be thinking again..... |
| 19:16.20 | qakhan | yes before |
| 19:16.26 | [TK]D-Fender | "A" is not right. But "B" is not right... let's try "A"! |
| 19:17.20 | qakhan | but it detect "Machine" with know that will someone accept call or not |
| 19:18.16 | [TK]D-Fender | qakhan: It is detecting your caller as a machine based on the rules it is given... |
| 19:19.54 | *** join/#asterisk navaismo (0c0adbdf@gateway/web/freenode/ip.12.10.219.223) |
| 19:20.22 | qakhan | ok but what about callee voicemail detection |
| 19:20.59 | [TK]D-Fender | qakhan: You clearly aren't doing it on the callee. |
| 19:22.28 | qakhan | i am sorry, i should have clear that i trying to detect callee voicemail |
| 19:24.23 | [TK]D-Fender | qakhan: And you should have used that knowledge to know where AMD would ahve to be run. |
| 19:24.26 | [TK]D-Fender | have* |
| 19:25.45 | *** join/#asterisk jhlavacek (~jirka@jix.nextadsl.cz) |
| 19:25.57 | qakhan | [TK]D-Fender please help me here |
| 19:26.17 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 19:26.19 | [TK]D-Fender | qakhan: Doing it BEFORE the Dial means you're doing it to the CALLER. |
| 19:26.43 | [TK]D-Fender | qakhan: Doing it AFTER the Dial means Dial has already exited for whatever reason and this is STILL the caller's channel |
| 19:26.48 | [TK]D-Fender | qakhan: What's left? |
| 19:27.53 | qakhan | i dont know |
| 19:28.52 | qakhan | so its mean we cannot detect VM? |
| 19:29.16 | [TK]D-Fender | qakhan: No. You're not thinking. |
| 19:29.35 | [TK]D-Fender | qakhan: Before Dial = bad. Afdter Dial = Bad. What's left? |
| 19:29.57 | qakhan | script |
| 19:30.00 | qakhan | :( |
| 19:30.09 | [TK]D-Fender | "script" doesn't mean anything |
| 19:30.13 | [TK]D-Fender | it is not a "time" |
| 19:31.05 | qakhan | i am blank here |
| 19:31.31 | [TK]D-Fender | That is a very bad sign |
| 19:31.42 | qakhan | yes you are right |
| 19:31.44 | [TK]D-Fender | You don't understand the very basic concepts of time. |
| 19:32.18 | navaismo | [TK]D-Fender: I've missed you so much, i mean, your bitter args |
| 19:32.57 | [TK]D-Fender | Oh, I'm not bitter. Just saddened at the state of several people. |
| 19:33.24 | [TK]D-Fender | Now if you said "futile args" .... then there'd be a point worthy of debate. |
| 19:33.28 | qakhan | navaismo [TK]D-Fender is really helpful guy |
| 19:33.55 | navaismo | I know |
| 19:34.31 | navaismo | and i mean it |
| 19:34.38 | navaismo | that was not a joke |
| 19:34.49 | navaismo | in fact i miss a lot people from here |
| 19:34.49 | qakhan | he try us to think about the problem and its sulotion that is why he does give strait answer :) |
| 19:35.21 | ChannelZ-Wk | It's nothing personal.. but I wish the people who invented these pre-dialing robocall things would spontaneously catch fire. |
| 19:35.22 | qakhan | see [TK]D-Fender how much i was buttering you :) |
| 19:35.24 | qakhan | hahhahahaha |
| 19:36.01 | navaismo | ChannelZ-Wk: me? O_o |
| 19:37.00 | [TK]D-Fender | DISCO INFERNO! |
| 19:37.07 | [TK]D-Fender | dances... |
| 19:37.22 | qakhan | so [TK]D-Fender your were talking about time |
| 19:38.04 | qakhan | wait for call to be pickedup in 25sec if not then consider as voice mail? |
| 19:38.20 | ChannelZ-Wk | Like I said, nothing personal. |
| 19:38.44 | [TK]D-Fender | qakhan: How does "no answer" equal "voicemail" in your opinion? |
| 19:43.39 | qakhan | yes good option |
| 19:44.29 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 19:44.29 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:45.31 | qakhan | but where i use ${DialStatus} to get the value |
| 19:45.40 | qakhan | "noanswer" |
| 19:46.03 | [TK]D-Fender | qakhan: that is AFTER the dial. |
| 19:46.30 | [TK]D-Fender | [14:26][TK]D-Fenderqakhan: Doing it AFTER the Dial means Dial has already exited for whatever reason and this is STILL the caller's channel |
| 19:46.37 | [TK]D-Fender | [14:29][TK]D-Fenderqakhan: Before Dial = bad. Afdter Dial = Bad. What's left? |
| 19:48.38 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 19:48.48 | qakhan | no idea, today i my mind is not working |
| 19:48.49 | *** join/#asterisk windback (~jkleinerm@190.123.122.162) |
| 19:52.59 | [TK]D-Fender | qakhan: I recommend going to sleep and coming back when you're in a healthy position to think about what you're oidng... or at least remember how time works. |
| 19:54.18 | *** join/#asterisk Ta^3 (~tacvbo@187.189.162.201) |
| 19:54.43 | windback | I'm receiving a call trough a DAHDI ISDN trunk. The incoming call hasn't the calling party part. When I try to set the caller id number using CALLERID function, it doesn't work and the call is received in my sip phon as Anonymous. Any idea why in this situation CALLERID function is not working? |
| 19:54.46 | windback | http://pastebin.com/8jcGeE7F |
| 19:55.51 | [TK]D-Fender | windback: because of a pricay setting on the inbound call. |
| 19:55.55 | [TK]D-Fender | privacy* |
| 19:56.15 | windback | [TK]D-Fender, is there any way to override it? |
| 19:56.26 | [TK]D-Fender | windback: Yes, there is a function for setting that... |
| 19:56.55 | windback | [TK]D-Fender, can you tell me how? |
| 19:57.57 | pukkita_ | <[TK]D-Fender>: the CONNECTEDLINE is a variable? a function? googling and trying but I don't understand exactly what it does |
| 19:58.23 | [TK]D-Fender | pukkita_: https://www.google.ca/#q=asterisk+CONNECTEDLINE |
| 19:58.41 | [TK]D-Fender | pukkita_: Funny, My google search for these 2 words tells me incredibly specifically what it is. |
| 19:59.38 | windback | I tried doing it using CONNECTEDLINE functionline with not success |
| 20:00.00 | [TK]D-Fender | windback: that wasn't for you... |
| 20:00.18 | [TK]D-Fender | windback: and that command clearly says nothing about "privacy" |
| 20:00.29 | windback | [TK]D-Fender, yes |
| 20:01.03 | [TK]D-Fender | windback: "core show function CALLERID" |
| 20:01.33 | [TK]D-Fender | windback: The ones you care about are "pres" |
| 20:01.36 | pukkita_ | I have voip-info, and asterisk wiki but I don't understand how it works |
| 20:01.53 | [TK]D-Fender | pukkita_: Let's see what you tried doing with it. |
| 20:02.10 | pukkita_ | hahahah |
| 20:02.13 | pukkita_ | :) |
| 20:02.34 | pukkita_ | I managed to screw up the whole system! |
| 20:02.49 | [TK]D-Fender | pukkita_: If you say so... |
| 20:03.23 | [TK]D-Fender | pukkita_: At least you find it funny. So you've got that going for you ... which is good. |
| 20:03.55 | pukkita_ | I'm trying that an agent receiving an answered call from an outgoing campaign gets the dialled number as callerid |
| 20:04.04 | navaismo | nice* |
| 20:04.12 | qakhan | [TK]D-Fender give me lil idea about time you are talking about |
| 20:04.39 | windback | [TK]D-Fender, Should i do something like Set(CALLERID(num-pres)=7777777) ?? |
| 20:04.55 | [TK]D-Fender | qakhan: Before, After, ........X. |
| 20:05.01 | [TK]D-Fender | qakhan: Fill in the blank. |
| 20:05.27 | [TK]D-Fender | windback: What does a search tell you you should set it to? |
| 20:05.58 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 20:06.40 | windback | [TK]D-Fender, I can't find information about CALLERID datatypes |
| 20:07.09 | [TK]D-Fender | windback: https://www.google.ca/#q=asterisk+CALLERID+num-pres |
| 20:07.28 | [TK]D-Fender | windback: My 2 second search with those keywords turned up plenty of instal results listing the allowed values. |
| 20:07.35 | [TK]D-Fender | instant* |
| 20:10.32 | qakhan | [TK]D-Fender are you talking about timeout in Dial()? |
| 20:11.12 | [TK]D-Fender | qakhan: I am looking to see if you have a basic understanding of time itself. So far it's really looking like "no" |
| 20:12.08 | qakhan | ok |
| 20:12.25 | qakhan | so what about time. is it a fuction or app? |
| 20:13.05 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 20:13.13 | [TK]D-Fender | qakhan: Time. Like on a clock. Like what we've wasted a lot of. |
| 20:13.50 | qakhan | :) |
| 20:13.55 | qakhan | hahaha |
| 20:14.32 | qakhan | it is related to callee voice mail detection? |
| 20:16.06 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 20:16.09 | pukkita_ | <[TK]D-Fender>what I don't understand is how CONNECTEDLINE works, is it initialized when the outgoing channel is created, then that channel carry that info to whatever other channels are handed to? |
| 20:16.34 | windback | [TK]D-Fender, Thank.. I could solve it doing: http://pastebin.com/T6xr8ads |
| 20:17.37 | pukkita_ | I initialized it specifically but when Verbose() prior to hand the call to the agent still shows our callerid |
| 20:19.01 | *** part/#asterisk lillian (~lillian@unaffiliated/lillian) |
| 20:19.25 | [TK]D-Fender | pukkita_: Maybe you did it wrong. |
| 20:19.40 | [TK]D-Fender | windback: Good.... |
| 20:23.22 | pukkita_ | A NoOP before leaving the initial context showed I did it right |
| 20:23.35 | pukkita_ | that's what puzzles me |
| 20:24.07 | [TK]D-Fender | pukkita_: We can't confirm this. |
| 20:24.08 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 20:26.23 | *** join/#asterisk fornax (~fornax@p5797AF3B.dip0.t-ipconnect.de) |
| 20:28.05 | qakhan | [TK]D-Fender still figuring out about time |
| 20:28.51 | newtonr | pukkita_, "what I don't understand is how CONNECTEDLINE works" <- http://tinyurl.com/ornvedm |
| 20:29.19 | windback | [TK]D-Fender, Thanks for your help and forgive me for my inability to search in google |
| 20:30.23 | newtonr | windback, helpful course on using Google: http://www.google.com/insidesearch/landing/powersearching.html |
| 20:30.24 | pukkita_ | newtonr: thanks, I have read that page a zillion times |
| 20:30.40 | pukkita_ | guess has to do with the phones |
| 20:30.58 | pukkita_ | trustrpid, sendrpid or whatever |
| 20:32.40 | pukkita_ | [What I don't understand is if its supposed to "bind" to a channel and carry that info to whoever other channels its bridged to |
| 20:34.02 | *** join/#asterisk BitEvil (~quassel@tor/regular/SpeedEvil) |
| 20:37.04 | *** join/#asterisk nny (~Scott@cpe-174-107-201-051.sc.res.rr.com) |
| 20:37.23 | nny | can someone explain this (Ast 11.6) No application 'RemoveQueueMemeber' for extension |
| 20:39.07 | nny | arg |
| 20:39.08 | nny | nm |
| 20:39.10 | nny | haha |
| 20:39.12 | nny | I am an idiot |
| 20:39.43 | nny | Memeber |
| 20:40.41 | [TK]D-Fender | nny: https://www.youtube.com/watch?v=OonDPGwAyfQ |
| 20:50.38 | *** part/#asterisk lvlinux (~lvlinux@unaffiliated/lvlinux) |
| 20:50.44 | *** join/#asterisk lvlinux (~lvlinux@unaffiliated/lvlinux) |
| 21:00.23 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:12.37 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:15.46 | Ta^3 | I just found in queue_log an exception in epoch time, a 13 digits value. |
| 21:15.52 | Ta^3 | 1415193520|NONE|NONE|Agent/2612|UNPAUSEALL| |
| 21:15.52 | Ta^3 | 1411415193742|NONE|NONE|Agent/2801|PAUSEALL| |
| 21:15.52 | Ta^3 | 1415193742|NONE|9024|Agent/2801|PAUSE| |
| 21:16.14 | Ta^3 | it's an asterisk 11.13.0 |
| 21:16.45 | Ta^3 | but i don't know how to dig deeper in that. |
| 21:17.28 | Ta^3 | i noticed because the queue_log parser i wrote complained with that |
| 21:17.31 | newtonr | Ta^3, has it only happened the one time? |
| 21:17.44 | Ta^3 | newtonr, yes, only that. |
| 21:17.49 | jvhester | <PROTECTED> |
| 21:17.59 | jvhester | 141 |
| 21:18.56 | Ta^3 | yes, double 141141 |
| 21:20.07 | Micc | Is there a way to make a digium phone show multiple line keys for the same account? |
| 21:20.18 | jvhester | ...I wonder if it could be reproduced with another PAUSEALL for that member Ta^3 |
| 21:22.37 | nny | POlycoms just might be the shittiest phones in existence |
| 21:22.54 | Ta^3 | jvhester, it's a busy server, there are others PAUSEALL for that member |
| 21:22.57 | Ta^3 | 1415194312|NONE|NONE|Agent/2801|UNPAUSEALL| |
| 21:22.57 | Ta^3 | 1415194312|NONE|9024|Agent/2801|UNPAUSE| |
| 21:23.01 | Ta^3 | withouth any issue |
| 21:24.35 | jvhester | that's an UNPAUSEALL |
| 21:26.00 | jvhester | But if you can reproduce it at will it may be easier to fix |
| 21:27.27 | newtonr | Ta^3, if you can reproduce it then you should definitely file an issue on issues.asterisk.org/jira |
| 21:27.39 | newtonr | if it just happened once, there is probably not much we can do to track it down. |
| 21:28.29 | Ta^3 | jvhester, newtonr thank you both, let me check logs, and see if i can reproduce it. |
| 21:33.40 | *** join/#asterisk Cynagen (~cynagen@ip72-208-60-104.ph.ph.cox.net) |
| 21:34.46 | nny | What is the proper way to configure a polycom to work behind NAT? My linksys phone doesn't require a damn static ip :\ |
| 21:35.42 | doop | nny: neither does my polycom |
| 21:36.21 | nny | doop must be reading an old article. Is there a specific XML entry to tell the phone it is behind NAT? |
| 21:37.01 | doop | nny: not that i remember. just tell it where your server is and what credentials it needs, then tell asterisk the phone is behind NAT |
| 21:37.20 | doop | have it grab an IP with dhcp when it boots, and you should be in business |
| 21:37.43 | nny | I am getting UNREACHABLE behind NAT right now, but not with my test soft phone |
| 21:38.12 | doop | maybe something's up with your gateway/firewall? tough to say at this point |
| 21:38.46 | nny | Nah just happens with Polycom. I hate them |
| 21:39.07 | doop | don't know what this "just happens" business is you're talking about, but it doesn't "just happen" with mine |
| 21:39.57 | nny | I can register another phone no problem from the same network, or a soft phone. The polycom doesn't work. |
| 21:40.29 | doop | how sure are you that you didn't fuck up the polycom configuration |
| 21:40.37 | nny | pretty fucking sure |
| 21:40.45 | nny | i can pastebin it |
| 21:41.28 | doop | if you want, sure |
| 21:42.39 | nny | http://pastebin.com/HuvNacgg |
| 21:42.51 | nny | both mac.cfg and sip-local.cfg there |
| 21:43.01 | nny | er local-sip.cfg |
| 21:43.08 | nny | they work if connected to the same network |
| 21:43.38 | doop | did you excise most of the sample configuration? |
| 21:43.46 | nny | yes |
| 21:43.57 | doop | that may be part of your issue |
| 21:44.16 | nny | may? |
| 21:44.46 | nny | I mean isn't there a way to see which line of the xml is used to configure that aspect? Does the phone default to awkward settings? |
| 21:44.50 | doop | have you tried altering the sample configuration to contain your config details without removing any of the provided file? |
| 21:45.07 | nny | why would I? |
| 21:45.12 | nny | sounds like magic |
| 21:45.29 | doop | try it and see if your results differ |
| 21:46.32 | nny | eh |
| 21:46.36 | nny | not how you troubleshoot |
| 21:46.40 | nny | so magic, got it |
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| 21:47.31 | doop | oh well |
| 21:47.50 | nny | even people who use polycoms aren't sure how they work heh |
| 21:47.55 | nny | \o/ |
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| 21:48.14 | doop | more along the lines of every aspect of the phone's behavior is controlled by those files |
| 21:48.25 | doop | if you remove most of it, the phone goes wtf |
| 21:48.58 | nny | multiple documentation on how to configure them says nothing about having every XMl line in the config |
| 21:49.03 | nny | XML* |
| 21:49.22 | doop | clearly you're more interested in arguing about it than just trying it |
| 21:49.24 | nny | if there is a setting for that aspect, sure I'd add it. But just shotgun approach does nothing |
| 21:49.28 | doop | good day sir |
| 21:49.35 | nny | no I am interested in which line changes it.. lol ok m'lady |
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| 21:55.04 | malcolmd | Micc: negative, we don't do multiple button appearances for calls. |
| 21:56.31 | nny | it appears the phone is choosing an arbitrary sip port to register on... |
| 21:56.55 | nny | and outside the network it doesn't have access to that port. Time to dig through the admin guide for that parameter |
| 21:57.02 | nny | \o/ polycom documentation |
| 22:02.25 | nny | fucking hell |
| 22:02.25 | nny | reg.1.outboundProxy.port="5060" |
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| 22:02.38 | nny | 10 different google search terms later |
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| 22:57.50 | *** join/#asterisk mic_ (~mic@0206701025.0.fullrate.dk) |
| 22:58.12 | mic_ | when can asterisk spit 603 Declined? |
| 22:59.58 | [TK]D-Fender | When it stops processing a call it didn't answer |
| 23:00.05 | [TK]D-Fender | and doesn't have anything else to say |
| 23:00.51 | mic_ | sec, parsing in my own head... |
| 23:02.46 | mic_ | [TK]D-Fender: what if the call was answered in AGI script? |
| 23:04.00 | mic_ | in the SIP trace I see asterisk sending 603 |
| 23:04.12 | mic_ | and in the log "auto fallthrough -> state unkown" |
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| 23:22.20 | [TK]D-Fender | [18:02]mic_[TK]D-Fender: what if the call was answered in AGI script? it wasn't |
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| 23:29.39 | mic_ | smells I found the problem. |
| 23:29.54 | mic_ | unfortunately, in-house spaghetti related. |
| 23:31.43 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 23:34.02 | mic_ | found |
| 23:34.04 | mic_ | reproduced. |
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