IRC log for #asterisk on 20170104

00:01.09Follow-meWIMPy ?
00:01.19Follow-measteriskmonkey any note
00:01.21Follow-mehere
00:01.38WIMPy?
00:02.02Follow-meso what else i need to do
00:02.09asteriskmonkeysorry not reading backlog
00:02.38*** join/#asterisk clopez (~tau@neutrino.es)
00:02.40WIMPyStart your extensions right, like I told you.
00:02.51Follow-meTo make caller A go to ringing mode while  i check after b answers if b is human bridge the calls if not hangup
00:12.50*** part/#asterisk kharwell (kharwell@nat/digium/x-gksdguftuvzwuxrc)
00:16.31[TK]D-FenderI don't see why you're using an Originate here at all
00:19.07*** join/#asterisk mahlon (~mahlon@martini.nu)
02:58.59*** join/#asterisk [[thufir]] (~thufir@192.157.116.163)
03:23.17*** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta)
04:00.17*** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com)
04:00.48*** join/#asterisk [[thufir]] (~thufir@192.157.116.163)
04:04.33*** join/#asterisk fstd_ (~fstd@unaffiliated/fisted)
04:13.41*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
04:14.34*** join/#asterisk [[thufir]] (~thufir@192.157.116.163)
04:21.29*** join/#asterisk Alblasco1702 (~Alblasco1@ip5456b46b.speed.planet.nl)
04:30.41*** join/#asterisk [[thufir]] (~thufir@192.157.116.163)
06:18.36*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
06:36.43*** join/#asterisk Follow-me (~Follow-me@212.34.23.43)
06:42.47Follow-meexten => _0X.,1,Originate(SIP/fras/${EXTEN},exten,zaid,1234,1)
06:43.13Follow-me[zaid]
06:43.13Follow-meexten => _0X.,n,AMD()
06:43.14Follow-meexten => _0X.,n,NoOp(${DIALSTATUS}) ; Here got ANSWER status
06:43.25Follow-meexten => _0X,n,ConfBridge(1234)
06:43.41Follow-mei need someone to help me with that
06:43.45Follow-meis it correct
06:43.47Follow-me?
06:52.29[TK]D-FenderCorrect for what?
06:52.44[TK]D-FenderYou can't just shove a bunch of dialplan lines and assume we know your expectation from it
06:54.13Follow-meshould place A  on ringing until in answered when answered it will jump to [zaid] context then will check AMD AND do bridge calls
06:54.28Follow-meuntill B is answered
06:54.43[TK]D-Fenderyou shouldn't talk about "A" so vaguely
06:55.07[TK]D-Fenderif ythe PHONE placing that call is supposed to ring... then Originate is the completely wrong approach
06:55.30[TK]D-FenderOriginate starts a completely separate call that is no way related to the dialplan executing
06:56.08Follow-mei have been reading all night the books and google
06:56.11[TK]D-Fenderand I alrewady told you what to do for this.
06:56.15[TK]D-FenderMACRO or GOSUBN
06:56.24[TK]D-Fender"core show applicastion dial"
06:56.35Follow-meso i shouldnt use originate
07:01.01Follow-meI need to place A on ringing while i check from B after B Answer i  need to run Amd to check if machine drop call if human bridge calls with A
07:01.32*** join/#asterisk bof22 (~Thunderbi@185.13.183.107)
07:02.51[TK]D-FenderThere is no "place on ringing;.
07:03.17Follow-mewhat will happen
07:03.20[TK]D-Fenderyou DIAL them and when they answer you run AMD AGAINST them to see if it is what you exect and resume if you like the result
07:03.21Follow-meto A
07:03.49Follow-mei cant put false ringing for A so i check B before open the line for A
07:04.15[TK]D-FenderYou don't need false ringing... you ARE ringing
07:04.41[TK]D-Fendercall isn't bridged until the macro completes without cause for abort
07:06.23Follow-meso i can make cause of abort if Amd returns machine right
07:07.00[TK]D-Fender<[TK]D-Fender> MACRO or GOSUB
07:07.08[TK]D-Fender[TK]D-Fender> "core show application dial"
07:21.59Follow-me[default]
07:21.59Follow-meexten => _0X,1,Macro(macro-zaid|${exten},s,1)
07:22.06Follow-meso it should be like this
07:22.13Follow-meand under [macro-zaid]
07:22.37Follow-meexten => _0X,1,Dial(SIP/${exten},20)
07:22.38Follow-meexten => n,AMD()
07:22.38Follow-meexten => n,GotoIf($[ "${AMDSTATUS}" = "HUMAN" ]?${EXTEN:1},1)
07:24.42[TK]D-FenderNO
07:24.50[TK]D-Fender[TK]D-Fender> <[TK]D-Fender> MACRO or GOSUB
07:24.50[TK]D-Fender<[TK]D-Fender> [TK]D-Fender> "core show application dial"
07:30.39Follow-meso i should use gosub or macro
07:36.38Follow-me[Default]
07:36.39Follow-meexten => _0X,1,zaid(${EXTEN})
07:36.39Follow-mesame => n ,Hangup()
07:37.00[TK]D-FenderWTF is that?
07:37.13[TK]D-FenderZaid is NOT a dialplan application and you are passing some random BS number
07:37.14Follow-me[zaid]
07:37.14Follow-meexten => _0X,1,Dial(${EXTEN})
07:37.14Follow-mesame => n ,AMD()
07:37.21[TK]D-Fenderwhat did you think that even meant>
07:37.30Follow-meexten => n,GotoIf($[ "${AMDSTATUS}" = "HUMAN" ]?${EXTEN:1},1)
07:37.30Follow-mesame=> n,return()
07:37.34[TK]D-FenderNO,
07:37.37[TK]D-FenderAll garbage
07:37.40Follow-mesorry
07:37.46[TK]D-Fendereven the dial is nonsense in any capacity
07:37.57[TK]D-Fender<[TK]D-Fender> [TK]D-Fender> <[TK]D-Fender> MACRO or GOSUB
07:37.57[TK]D-Fender<[TK]D-Fender> <[TK]D-Fender> [TK]D-Fender> "core show application dial"
07:44.02*** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net)
07:46.17Follow-meso dial should be executed then gosub right
07:47.08[TK]D-FenderREAD DIAL'S FUCKING INSTRUCTIONS.  HOW MANY TIMES DO I HAVE TO SAY IT?
07:47.13Follow-meor dial should be in context
07:47.28[TK]D-Fender<PROTECTED>
07:47.33[TK]D-FenderREAD THE DAMN INSTRUCTIONS
07:47.40[TK]D-Fender[TK]D-Fender> <[TK]D-Fender> [TK]D-Fender> <[TK]D-Fender> MACRO or GOSUB
07:47.41[TK]D-Fender^^^^^
07:47.52[TK]D-FenderPARAMETERS TO THE DAMN DIAL COMMAND
07:48.18[TK]D-FenderNothing AFTER a Dial means ANYTHING
07:48.39[TK]D-Fenderif that next line calls it's because the dial ABORTED having not been answered
07:48.47[TK]D-FenderDial is a BLOCKING app
07:48.52[TK]D-Fenderlike pretty much all of them
07:54.53Follow-megosub wull jump to the context but we should do dial before so when we put our condition in gosub ther ewill be results
07:55.08Follow-meon the websites it show gosub is executed before dial
07:55.19[TK]D-FenderREAD TEH FUCKING INSTRUCTIONS
07:55.20Follow-mei have read the core show application dialplan
07:55.39[TK]D-FenderGosub/Macro is executed on the CALLED CHANNEL before bridge
07:55.50[TK]D-FenderThis is NOT ***before*** dial
07:56.11Follow-meso gosub or macro both works
08:00.23*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)
08:06.34Follow-meok now [default]
08:06.38Follow-meexten => _0X,1,Gosub(check)
08:06.38Follow-meexten => _0X,n,Dial(SIP/${EXTEN})
08:06.41[TK]D-FenderNO
08:06.47Follow-me[check]
08:06.47Follow-meexten => _0X,n,AMD()
08:06.48Follow-meexten => _0X,n,GotoIf($[ "${AMDSTATUS}" = "Machine" ]?Hangup
08:06.50[TK]D-FenderNO
08:06.53Follow-meexten=> _0X,n,Return()
08:07.13[TK]D-Fender**************NO*********
08:07.23Follow-me:(
08:07.35Follow-mewhats wrong
08:07.36[TK]D-Fender<Follow-me> exten => _0X,n,AMD() <- what part of NON of this shit gets called AFTER Dial unless it has FAILED are you having trouble with?
08:07.47[TK]D-Fender<PROTECTED>
08:07.54[TK]D-FenderYou are doing the SAME stupid thing every pasted
08:08.06[TK]D-FenderI told youi top use one of theose 2 FUCKING OPTIONS IN THE DIAL COMMAND
08:08.11[TK]D-FenderREAD THE INSTRUCITONS
08:11.12[TK]D-Fenderis done for the night
08:13.07*** join/#asterisk mirela666 (~mirkob@2a00:1950:400:0:85e0:32c7:3ae4:2ae8)
08:17.02*** join/#asterisk [[thufir]] (~thufir@192.157.116.163)
08:18.26*** join/#asterisk pchero_work (~pchero@2a00:c80:1072::159)
08:18.42*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:18.47igcewielingheh, Google Play Games suggested "RippedCucumber1830" for a username.  I wonder if that is anything like an eggplant...
08:24.42*** join/#asterisk AndyCap (~aoy@pdpc/supporter/sustaining/AndyCap)
08:26.09Follow-me[default]
08:26.09Follow-meexten => _0X,1,Gosub(check,_0X,1,${EXTEN})
08:26.20Follow-me[check]
08:26.20Follow-meexten => start,1,NoOp()
08:26.21Follow-mesame => n,Dial(${EXTEN},10)
08:26.21Follow-mesame=> n,AMD()
08:26.28Follow-mesame => n,GotoIf($["${AMDSTATUS}" = "MACHINE"]?busy:unavail)
08:26.52Follow-meIS THIS CORRECT ?
08:31.24Follow-mehello
08:31.39Follow-meanybody have time to look at this please
08:35.42*** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl)
08:36.45samwieremaGood morning, is there a way to reload the ARI HTTP server (from the cli) without restarting Asterisk?
08:40.19*** join/#asterisk Y04NN (~y04nn@178.18.54.206)
08:46.24*** join/#asterisk mirela666 (~mirkob@2a00:1950:400:0:2147:a474:c892:bb8)
08:50.30*** join/#asterisk Y04NN (~y04nn@178.18.54.206)
09:15.59Follow-mewhere is the location after B is answered
09:17.23*** join/#asterisk mirela666 (~mirkob@2a00:1950:400:0:c9b8:15f8:5ff:1218)
09:17.50*** join/#asterisk sekil (~sekil@cable-89-216-231-150.dynamic.sbb.rs)
09:26.58*** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za)
09:29.52*** join/#asterisk pawiecki (~pawiecki@router.dir.pl)
09:32.06pawieckiI have one SIP Trunk registered, but my Asterisk doesn't respond to INVITE. I've done packet capture, but I don't see any response to Invites. It happened overnight, when no changes were made. What could be the problem here?
09:33.31pawieckiYesterday it worked fine, with no problems. I believe it's not router/network's fault, because I captured files on the * box.
09:33.41pawieckipackets*
09:42.35*** join/#asterisk mahlon (~mahlon@martini.nu)
09:54.59*** join/#asterisk JBLITX (~jbl@189.11.57.50)
10:05.51*** join/#asterisk cryptic_ (~cryptic@67-8-35-31.res.bhn.net)
10:19.37*** join/#asterisk stefan27 (~stefan27@static-212-247-4-149.cust.tele2.se)
10:20.28*** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta)
10:22.50stefan27In order to receive fax on a sip channel with ReceiveFAX() , do I need licenses for T38 or is the module res_fax_spandsp.so sufficient?
10:23.59stefan27"sip show peer bla" reports T.38 support: Yes, does that refer to only pass-through support?
10:35.57Follow-mei need someone to help me with DIal
10:36.45*** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es)
10:37.00pawieckiFollow-me: just describe your problem...
10:39.09Follow-me[zaid]
10:39.09Follow-meexten => start,1,Dial(SIP/${EXTEN},10)
10:39.24Follow-me[default]
10:39.27Follow-meexten => _0X.,1,GoSub(zaid,start,1(${EXTEN})
10:39.51Follow-methis should send number and let context zaid dial it right ?
10:40.14Follow-meapp_dial.c:2375 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
10:41.14Guggethe verbose log should show you what is wrong
10:41.28pawieckiFollow-me: looks like your SIP peer is not registered
10:42.18Follow-meBasically what im tryin to do is when A Calls B i jump to [zaid] and want to make dial after B answers i want to do check if its human return call if machine hangup
10:42.38Follow-mei want the call to be out from provider called fras
10:43.04ckaFollow-me: If an endpoint in the zaid context dials start, asterisk will translate the line to Dial(SIP/start,10) is that what you want?
10:45.32Follow-mei want it to DIAL SIP FROM TRUNK ( PROVIDER fras) and after answer do checks then connect with A
10:46.04ckaFollow-me: Sounds like you want to run some dialplan on the called channel? Maybe the U option to dial, will do that?
10:46.23Follow-meyes
10:47.02ckaFollow-me: I guess you should look at either the G or U option to dial.
10:47.53Follow-mewhere is that
10:48.38ckaFollow-me: Or maybe the M option..
10:49.09ckaFollow-me: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial Theres some samples there of using options with dial?
10:49.43ckaFollow-me: You would most likely be able to use the sample titles "Example: Dial with post-answer subroutine executed on outbound channel
10:49.50ckaFollow-me: You would most likely be able to use the sample titles "Example: Dial with post-answer subroutine executed on outbound channel" as a starting point
10:52.06Follow-me<PROTECTED>
10:52.07Follow-me<PROTECTED>
10:52.07Follow-me[2017-01-04 11:51:45] ERROR[30439]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("start", "(null)", ...): Name or service not known
10:52.20Follow-me[2017-01-04 11:51:45] WARNING[30439]: chan_sip.c:5713 create_addr: No such host: start
10:52.20Follow-me[2017-01-04 11:51:45] WARNING[30439]: app_dial.c:2375 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
10:53.28ckaFollow-me: You don't have a sip target with the hostname "start"? Then you need to change it to call the actual number you want...
11:00.53*** join/#asterisk netman (~netman@185.94.249.77)
11:08.53Follow-me<PROTECTED>
11:08.53Follow-me[2017-01-04 12:08:42] ERROR[9177]: app_stack.c:559 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:mor_local, Extension:zaid, Priority:1)
11:28.06*** join/#asterisk follow-me (~Follow-me@212.34.23.94)
11:28.19follow-me<PROTECTED>
11:28.19follow-me<Follow-me> [2017-01-04 12:08:42] ERROR[9177]: app_stack.c:559 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:mor_local, Extension:zaid, Priority:1)
11:37.41follow-mehello
11:38.11*** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou)
11:41.02*** join/#asterisk mirela666 (~mirkob@89.184.168.160)
11:46.23follow-mecka
11:46.24follow-mehello
12:00.05defsworkanyone used Cisco 8800 handsets ?  trying to find out if there is a way to get the .xml config off it
12:01.50*** join/#asterisk Y04NN (~y04nn@178.18.54.206)
12:26.21*** join/#asterisk blinky_ (~damia@host81-136-203-112.in-addr.btopenworld.com)
12:27.08blinky_Morning all, I am wondering if someone could point me in the direction of some free monitoring software that may help me diagnose call quality issues we are having at work?
12:30.19*** join/#asterisk tuxian (~tuxian@194.12.3.78)
12:30.21pawieckiblinky_: other than Wireshark?
12:30.44follow-mehello
12:30.51pawieckifollow-me: hi
12:31.01follow-meanybody here knpw how to do bridged call with originate
12:31.23blinky_pawiecki, didn't think of wireshark, will have a look now.
12:32.19pawieckiblinky_: I mainly use tcpdump to capture packets on linux server, then analyze it in Wireshark. It's very usefull tool.
12:33.51blinky_pawiecki, I will run it now, am new to elastix and have been asked to see if I can figure out what is going on, I have eliminated an external issue so now looking at testing internally.
12:34.49blinky_brb got to reboot
12:39.03*** join/#asterisk blinky_ (~damia@host81-136-203-112.in-addr.btopenworld.com)
12:41.35ckafollow-me: I get the feeling, that you are missing some of the basics of how asterisk and its dialplan works. And that is more than I have the time to teach you, unfortunatly..
12:41.46follow-me<PROTECTED>
12:50.10pawieckifollow-me: try to make things simpler. First, try to make a simple dial, to another extension. Then step by step, try to understand how it works.
12:52.57*** join/#asterisk Dpunkt (~Dpunkt@p54945B59.dip0.t-ipconnect.de)
13:01.47follow-meI ALREADY did
13:01.56follow-meand im using amd for outgoing
13:03.27*** join/#asterisk sekil (~sekil@cable-89-216-231-150.dynamic.sbb.rs)
13:05.08follow-meapp_dial.c:2375 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
13:05.16follow-mewhere do i place subscriber in dial
13:05.30follow-meDial("SIP/ipauth2xMJEKBj-00000005", "SIP/000201277682533,,U(default^called_channel^1),30") in new stack
13:06.18pawieckifollow-me: go to your Asterisk CLI, then 'core show application dial' and check if your syntax is correct
13:06.30*** join/#asterisk tuxian_ (~tuxian@194.12.3.78)
13:23.14blinky_What is classed as unacceptable jitter? I have 1.359ms at the moment on one call?
13:26.19stefan27which function did you use to get data data (1.359 ms)?
13:26.23stefan27that data*
13:26.47blinky_It was found using tcpdump and looking at the data with wireshark
13:28.18stefan27All right, I have a related question on what unit is reported by asterisk's cli command "sip show channelstats" at column "Jitter"
13:29.40blinky_bare with me I will check
13:29.53stefan27I haven't looked at the mathematical definition(s) of jitter, so I wouldn't know
13:30.33blinky_At the moment I have no one in an active call, I will have to wait for someone to make a call
13:38.57stefan27I'm trying to find a good definition of jitter. It seems to me you should specify which audio-direction you are calculating the jitter in. I would guess jitter is defined as some function of the sequence of timestamps of received audio packets, where the jitter would be zero if timestamp(i+1)=timestamp(i)+20 ms in the case of standard call
13:53.30*** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com)
13:53.32*** join/#asterisk dascodemonkey (18f7ef72@gateway/web/freenode/ip.24.247.239.114)
13:54.17blinky_stefan27, similar to what I found - "Jitter - a variation in packet transit delay caused by queuing, congestion, timing drifts, route changes and serialization effects on the path through the network; the maximum allowable duration of jitter is 40 ms before deterioration occurs."
13:56.51*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
13:58.00dascodemonkeyI'm looking for a way to have incoming calls get categorized via a field in the cdr data so I can see how many sales calls came in for example. When the call comes in the agent that answers the call would push a button/enter a code on their phone during the call to append this data to the cdr record. Can someone point me in the right direction?
14:00.33SamotHow do they get to sales now?
14:06.32[TK]D-Fenderdascodemonkey, several different approaches possible for this
14:07.26dascodemonkeyAll calls come in via an ivr to multiple queues, but we get a lot of callers that must just smash the keypad at the ivr cause we get sales in support and customer service in sales... I'm just looking for a way to get more accurate cdr data. It's the same group of people answering, we're wanting to remove the ivr and go to one queue and have the agents categorize the call.
14:08.32dascodemonkeyD-fender do you know of any examples I could reference?
14:10.20[TK]D-Fenderdascodemonkey, When calling the agent you can for instance log the uniqueid and the agent being called so that the agent can call another extension that pulls that value associated and update either an existing field like authcode, or you could store it in a parallel table for linking
14:10.43follow-me<[TK]D-Fender> welcome back sir
14:11.27follow-me[zaid]
14:11.27follow-meexten => start,1,Dial(SIP/${EXTEN},10)
14:11.31yangIs there some "second hand gear" European VoIP forum, where I could sell VoIP stuff ?
14:11.35[TK]D-FenderYou aren't listeing
14:11.39follow-me[default]
14:11.39follow-me<PROTECTED>
14:11.39[TK]D-FenderI rtold you to use a DIAL OPTION
14:11.44[TK]D-Fenderand you still aren't
14:11.47follow-me<PROTECTED>
14:11.49follow-mebut still same
14:12.09[TK]D-Fender<follow-me>  exten => _0X.,1,GoSub(zaid,start,1(${EXTEN}) <- never use this dialplan application.
14:12.19[TK]D-FenderUSE THE DIAL OPTION ONLY
14:12.20[TK]D-FenderNOTHING ELSE
14:13.20follow-medial option should be in [default
14:13.24pawiecki[TK]D-Fender: do you mean GoSub? What's wrong with it?
14:14.00[TK]D-Fenderpawiecki, he should nott be using tthis
14:14.21[TK]D-FenderI wasted HOURS on this garbage when he can'tt use ONE STUPID DIAL PARAMETER
14:14.49[TK]D-Fenderfollow-me, I told you 100 times to DIAL the stupid target and pass a GOSUB or MACRO DIAL OPTION
14:15.25[TK]D-FenderRead Dial()'s INSTRUCTIONS and put the DIAL OPTION for it
14:16.22[TK]D-Fenderyang, https://www.craigslist.ca/about/sites#EU
14:17.55*** join/#asterisk somepoortech (~somepoort@72-0-128-68.static.firstlight.net)
14:18.41yang[TK]D-Fender: I thought about VoIP-specific forums...
14:19.20follow-meim afraid im using kolmisoft asterisk
14:19.22follow-mecompiled
14:19.28follow-methats why it didnt work i think
14:19.35[TK]D-Fenderyang, In North America we've got Kijiji & Craigslist.  Being in a major city there is a huge amount of stuff available
14:19.38follow-medial in mor_app
14:19.56[TK]D-FenderAs for "some web forum", no idea.  I'm wondering how anyone gets tto finding one so specific to voip.
14:20.18yangok
14:20.50yang[TK]D-Fender: basically want to get rid of the older VoIP telephones and eventually upgrade to Polycom VVX like you suggested.
14:21.23[TK]D-Fenderfollow-me, You are nor following basic instructions.  I told you to put shit IN THE DIAL COMMAND 100 times and you are still here wasting everyone's time
14:22.15[TK]D-Fenderyang, I'm unaware of resale sites & services for EU unfortunately.  I suppose there's always ebay...
14:22.23pawieckiIf I send REGISTER from * host A (from IP 'X') to host B (to IP 'Y'), and I got response 200 OK from host B (from IP 'Y') to host A (to IP 'Z') - Asterisk behind NAT; X - * internal IP; Y - sip trunk host; Z is * external IP - then should it register properly?
14:22.43sekilhi
14:23.26yang[TK]D-Fender: There is EBAY, true
14:24.50[TK]D-Fenderpawiecki, pawiecki if A is NAT'd then the packet should be sourced with the NAT address
14:24.51sekilpawiecki: if you got 200 OK should be ok
14:25.08[TK]D-Fendersekil, Not if * can't match it to the request
14:25.21Samotfollow-me: MOR != Asterisk. I thought you said you were switching from one to the other?
14:25.43sekilpawiecki: although I've seen providers sending 200 OK with additional X-headers which say that the money is zero
14:26.25sekil[TK]D-Fender: I meant he got the 200 OK back to his internal *
14:26.59[TK]D-Fendersekil, We're gettting implications of it coming back to a different IP that it was sent from
14:27.37sekil[TK]D-Fender: yeah..nat issue...
14:28.06[TK]D-Fenderbut I'm not seeing debug for this
14:28.09[TK]D-Fenderwhich we SHOULD
14:28.16sekilpawiecki: I guess if you have static ip you can use externip
14:28.21pawiecki[TK]D-Fender: right now it's not registering - timeouts, that's wy I asked. Packet is delivered to *, but the IP has changed
14:28.26[TK]D-Fenderbecause tracking somebody's re-written story gets convoluted for nothing
14:28.29sekilpawiecki: or you're on dynamic..you can use some stun server
14:29.20[TK]D-Fenderpawiecki, Show the actual debug
14:29.38pawiecki[TK]D-Fender: ok, please wait a bit.
14:29.41sekilpawiecki: which IP has changed?
14:30.37sekilpawiecki: on the router?
14:31.46sekilhates nat
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16:40.46igcewielingsekil: weird, it has been years since I've seen a major NAT issue with Asterisk.
16:42.31igcewielingWe generally don't allow our users to be on dynamic IPs, however.
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20:20.20ctjctjHello.  I'm writing some AMI code.  I want to know what buttons somebody presses when a call reaches them.  I start with an "Originate" and I see a sequence of events that takes place there after.  I am unsure how to associate those events with the Originate that I initiated.
20:21.46ctjctjThink "This is example.com calling to verify that this is the correct phone number.  Please press 1 this is your phone, press 2 to be added to our do not call list"
20:22.05igcewielingyou would do that on the other side of the originate
20:22.29ctjctjigcewieling, I don't follow.  What is the "other side of the originate?"
20:22.44igcewielingctjctj: what is your originate command?
20:23.07igcewielingI can explain in the terms of .call files, which are similar enough it might be applicable.
20:24.40ctjctjChannel = "SIP/trunk/4435551212", Context="outbound" Exten=4321, CallerID="example.com <201-555-1212>", Variable= (stuff), Priority = 1)
20:25.16igcewielingyou would playback the message in exten => 4321 section, usually using playback and read
20:25.30ctjctj*nods* Yes.
20:26.30[TK]D-FenderSo HOW do you want to log those?
20:26.45[TK]D-FenderDo you want to use AMI to listen for the logging?  You could do it direct in the dialplan as well.
20:27.07[TK]D-FenderDirect to file, DB, etc.  Or shove our AMI messages for VIR selections, etc
20:27.10[TK]D-FenderIVR*
20:27.17fileno no, not direct to me
20:27.35ctjctj[TK]D-Fender, I want to get an event that says "read 1" or "read 2".  I see events flowing, but I don't see how to associate the originate with the events.
20:27.58ctjctjThe events have Uniqueid, but I don't see how to map it to an ActionID
20:28.10igcewielingyou could collect the digits with READ, then generate a manager or CEL event with the info.
20:28.14[TK]D-Fenderstuff file
20:28.15[TK]D-Fender:p
20:28.52[TK]D-Fenderctjctj, When you originate a channel you can a CHANNEL variable to uniquely make tthem identifiable
20:29.03wyounghappy new year to you [TK]D-Fender
20:29.06[TK]D-Fenderctjctj, And use that witth sending a custom event
20:29.26[TK]D-Fender<ctjctj> The events have Uniqueid, but I don't see how to map it to an ActionID <- shove THAT intto a channel var
20:31.06ctjctj[TK]D-Fender, let me replay that: I generate a "unique id".  Use that for both the ActionID and the a Channel Variable.  Watch for the event that says setvar, and then do the map from that?
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20:55.11dmasieroAnyone know of a device that will stream on-premise hold music back to a hosted Asterisk PBX?
21:09.41SamotOne that will act as a streaming server for Asterisk to be a client for?
21:10.41igcewielingA HOSTED Asterisk?
21:11.39SamotI'm guessing this is going to stream over the Internet.
21:11.51dmasieroAt client location is a device that generates music, has a 3.5 mini jack out. Client's PBX is Asterisk in the cloud.
21:12.15SamotIf you can't attach a physical device to the PBX
21:12.37SamotYou'll have to do some sort of streaming client/server setup
21:13.44dmasieroWas wondering if anyone has ever come across a device that'll accomplish that task. Essentially, streaming that audio back to the PBX via a SIP channel.
21:15.29SamotIt wouldn't be a SIP channel.
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21:21.44SamotYou have to setup the musiconhold.conf to call on an application like mpeg123 that can stream from url..
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22:36.04igcewielingasterisk's moh can handle streaming sources, the problem is that, as far as I know, if the stream is interrupted it won't restart automatically.   That could have been fixed in more recent versions.
22:38.40igcewielingHave you looked in musiconhold.conf.sample?  You should.
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23:11.10SamotI think the real issue is the MoH unit is analog and doesn't do streaming.
23:11.20SamotSo they would need a completely new unit or a service this does this.
23:11.52SamotAnd hopefully a decent connection.
23:12.01SamotPhones, Data, MoH streaming...
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