00:01.09 | Follow-me | WIMPy ? |
00:01.19 | Follow-me | asteriskmonkey any note |
00:01.21 | Follow-me | here |
00:01.38 | WIMPy | ? |
00:02.02 | Follow-me | so what else i need to do |
00:02.09 | asteriskmonkey | sorry not reading backlog |
00:02.38 | *** join/#asterisk clopez (~tau@neutrino.es) |
00:02.40 | WIMPy | Start your extensions right, like I told you. |
00:02.51 | Follow-me | To 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-Fender | I 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) |
06:42.47 | Follow-me | exten => _0X.,1,Originate(SIP/fras/${EXTEN},exten,zaid,1234,1) |
06:43.13 | Follow-me | [zaid] |
06:43.13 | Follow-me | exten => _0X.,n,AMD() |
06:43.14 | Follow-me | exten => _0X.,n,NoOp(${DIALSTATUS}) ; Here got ANSWER status |
06:43.25 | Follow-me | exten => _0X,n,ConfBridge(1234) |
06:43.41 | Follow-me | i need someone to help me with that |
06:43.45 | Follow-me | is it correct |
06:43.47 | Follow-me | ? |
06:52.29 | [TK]D-Fender | Correct for what? |
06:52.44 | [TK]D-Fender | You can't just shove a bunch of dialplan lines and assume we know your expectation from it |
06:54.13 | Follow-me | should place A on ringing until in answered when answered it will jump to [zaid] context then will check AMD AND do bridge calls |
06:54.28 | Follow-me | untill B is answered |
06:54.43 | [TK]D-Fender | you shouldn't talk about "A" so vaguely |
06:55.07 | [TK]D-Fender | if ythe PHONE placing that call is supposed to ring... then Originate is the completely wrong approach |
06:55.30 | [TK]D-Fender | Originate starts a completely separate call that is no way related to the dialplan executing |
06:56.08 | Follow-me | i have been reading all night the books and google |
06:56.11 | [TK]D-Fender | and I alrewady told you what to do for this. |
06:56.15 | [TK]D-Fender | MACRO or GOSUBN |
06:56.24 | [TK]D-Fender | "core show applicastion dial" |
06:56.35 | Follow-me | so i shouldnt use originate |
07:01.01 | Follow-me | I 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-Fender | There is no "place on ringing;. |
07:03.17 | Follow-me | what will happen |
07:03.20 | [TK]D-Fender | you 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 |
07:03.21 | Follow-me | to A |
07:03.49 | Follow-me | i cant put false ringing for A so i check B before open the line for A |
07:04.15 | [TK]D-Fender | You don't need false ringing... you ARE ringing |
07:04.41 | [TK]D-Fender | call isn't bridged until the macro completes without cause for abort |
07:06.23 | Follow-me | so 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" |
07:21.59 | Follow-me | [default] |
07:21.59 | Follow-me | exten => _0X,1,Macro(macro-zaid|${exten},s,1) |
07:22.06 | Follow-me | so it should be like this |
07:22.13 | Follow-me | and under [macro-zaid] |
07:22.37 | Follow-me | exten => _0X,1,Dial(SIP/${exten},20) |
07:22.38 | Follow-me | exten => n,AMD() |
07:22.38 | Follow-me | exten => n,GotoIf($[ "${AMDSTATUS}" = "HUMAN" ]?${EXTEN:1},1) |
07:24.42 | [TK]D-Fender | NO |
07:24.50 | [TK]D-Fender | [TK]D-Fender> <[TK]D-Fender> MACRO or GOSUB |
07:24.50 | [TK]D-Fender | <[TK]D-Fender> [TK]D-Fender> "core show application dial" |
07:30.39 | Follow-me | so i should use gosub or macro |
07:36.38 | Follow-me | [Default] |
07:36.39 | Follow-me | exten => _0X,1,zaid(${EXTEN}) |
07:36.39 | Follow-me | same => n ,Hangup() |
07:37.00 | [TK]D-Fender | WTF is that? |
07:37.13 | [TK]D-Fender | Zaid is NOT a dialplan application and you are passing some random BS number |
07:37.14 | Follow-me | [zaid] |
07:37.14 | Follow-me | exten => _0X,1,Dial(${EXTEN}) |
07:37.14 | Follow-me | same => n ,AMD() |
07:37.21 | [TK]D-Fender | what did you think that even meant> |
07:37.30 | Follow-me | exten => n,GotoIf($[ "${AMDSTATUS}" = "HUMAN" ]?${EXTEN:1},1) |
07:37.30 | Follow-me | same=> n,return() |
07:37.34 | [TK]D-Fender | NO, |
07:37.37 | [TK]D-Fender | All garbage |
07:37.40 | Follow-me | sorry |
07:37.46 | [TK]D-Fender | even 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) |
07:46.17 | Follow-me | so dial should be executed then gosub right |
07:47.08 | [TK]D-Fender | READ DIAL'S FUCKING INSTRUCTIONS. HOW MANY TIMES DO I HAVE TO SAY IT? |
07:47.13 | Follow-me | or dial should be in context |
07:47.28 | [TK]D-Fender | <PROTECTED> |
07:47.33 | [TK]D-Fender | READ 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-Fender | PARAMETERS TO THE DAMN DIAL COMMAND |
07:48.18 | [TK]D-Fender | Nothing AFTER a Dial means ANYTHING |
07:48.39 | [TK]D-Fender | if that next line calls it's because the dial ABORTED having not been answered |
07:48.47 | [TK]D-Fender | Dial is a BLOCKING app |
07:48.52 | [TK]D-Fender | like pretty much all of them |
07:54.53 | Follow-me | gosub wull jump to the context but we should do dial before so when we put our condition in gosub ther ewill be results |
07:55.08 | Follow-me | on the websites it show gosub is executed before dial |
07:55.19 | [TK]D-Fender | READ TEH FUCKING INSTRUCTIONS |
07:55.20 | Follow-me | i have read the core show application dialplan |
07:55.39 | [TK]D-Fender | Gosub/Macro is executed on the CALLED CHANNEL before bridge |
07:55.50 | [TK]D-Fender | This is NOT ***before*** dial |
07:56.11 | Follow-me | so gosub or macro both works |
08:00.23 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
08:06.34 | Follow-me | ok now [default] |
08:06.38 | Follow-me | exten => _0X,1,Gosub(check) |
08:06.38 | Follow-me | exten => _0X,n,Dial(SIP/${EXTEN}) |
08:06.41 | [TK]D-Fender | NO |
08:06.47 | Follow-me | [check] |
08:06.47 | Follow-me | exten => _0X,n,AMD() |
08:06.48 | Follow-me | exten => _0X,n,GotoIf($[ "${AMDSTATUS}" = "Machine" ]?Hangup |
08:06.50 | [TK]D-Fender | NO |
08:06.53 | Follow-me | exten=> _0X,n,Return() |
08:07.13 | [TK]D-Fender | **************NO********* |
08:07.23 | Follow-me | :( |
08:07.35 | Follow-me | whats wrong |
08:07.36 | [TK]D-Fender | <Follow-me> exten => _0X,n,AMD() <- what part of NON of this shit gets called AFTER Dial unless it has FAILED are you having trouble with? |
08:07.47 | [TK]D-Fender | <PROTECTED> |
08:07.54 | [TK]D-Fender | You are doing the SAME stupid thing every pasted |
08:08.06 | [TK]D-Fender | I told youi top use one of theose 2 FUCKING OPTIONS IN THE DIAL COMMAND |
08:08.11 | [TK]D-Fender | READ THE INSTRUCITONS |
08:11.12 | [TK]D-Fender | is done for the night |
08:13.07 | *** join/#asterisk mirela666 (~mirkob@2a00:1950:400:0:85e0:32c7:3ae4:2ae8) |
08:17.02 | *** join/#asterisk [[thufir]] (~thufir@192.157.116.163) |
08:18.26 | *** join/#asterisk pchero_work (~pchero@2a00:c80:1072::159) |
08:18.42 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
08:18.47 | igcewieling | heh, Google Play Games suggested "RippedCucumber1830" for a username. I wonder if that is anything like an eggplant... |
08:24.42 | *** join/#asterisk AndyCap (~aoy@pdpc/supporter/sustaining/AndyCap) |
08:26.09 | Follow-me | [default] |
08:26.09 | Follow-me | exten => _0X,1,Gosub(check,_0X,1,${EXTEN}) |
08:26.20 | Follow-me | [check] |
08:26.20 | Follow-me | exten => start,1,NoOp() |
08:26.21 | Follow-me | same => n,Dial(${EXTEN},10) |
08:26.21 | Follow-me | same=> n,AMD() |
08:26.28 | Follow-me | same => n,GotoIf($["${AMDSTATUS}" = "MACHINE"]?busy:unavail) |
08:26.52 | Follow-me | IS THIS CORRECT ? |
08:31.24 | Follow-me | hello |
08:31.39 | Follow-me | anybody have time to look at this please |
08:35.42 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
08:36.45 | samwierema | Good 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) |
09:15.59 | Follow-me | where 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) |
09:32.06 | pawiecki | I 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? |
09:33.31 | pawiecki | Yesterday it worked fine, with no problems. I believe it's not router/network's fault, because I captured files on the * box. |
09:33.41 | pawiecki | packets* |
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) |
10:22.50 | stefan27 | In 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? |
10:23.59 | stefan27 | "sip show peer bla" reports T.38 support: Yes, does that refer to only pass-through support? |
10:35.57 | Follow-me | i need someone to help me with DIal |
10:36.45 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
10:37.00 | pawiecki | Follow-me: just describe your problem... |
10:39.09 | Follow-me | [zaid] |
10:39.09 | Follow-me | exten => start,1,Dial(SIP/${EXTEN},10) |
10:39.24 | Follow-me | [default] |
10:39.27 | Follow-me | exten => _0X.,1,GoSub(zaid,start,1(${EXTEN}) |
10:39.51 | Follow-me | this should send number and let context zaid dial it right ? |
10:40.14 | Follow-me | app_dial.c:2375 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
10:41.14 | Gugge | the verbose log should show you what is wrong |
10:41.28 | pawiecki | Follow-me: looks like your SIP peer is not registered |
10:42.18 | Follow-me | Basically 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 |
10:42.38 | Follow-me | i want the call to be out from provider called fras |
10:43.04 | cka | Follow-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? |
10:45.32 | Follow-me | i want it to DIAL SIP FROM TRUNK ( PROVIDER fras) and after answer do checks then connect with A |
10:46.04 | cka | Follow-me: Sounds like you want to run some dialplan on the called channel? Maybe the U option to dial, will do that? |
10:46.23 | Follow-me | yes |
10:47.02 | cka | Follow-me: I guess you should look at either the G or U option to dial. |
10:47.53 | Follow-me | where is that |
10:48.38 | cka | Follow-me: Or maybe the M option.. |
10:49.09 | cka | Follow-me: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial Theres some samples there of using options with dial? |
10:49.43 | cka | Follow-me: You would most likely be able to use the sample titles "Example: Dial with post-answer subroutine executed on outbound channel |
10:49.50 | cka | Follow-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 |
10:52.06 | Follow-me | <PROTECTED> |
10:52.07 | Follow-me | <PROTECTED> |
10:52.07 | Follow-me | [2017-01-04 11:51:45] ERROR[30439]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("start", "(null)", ...): Name or service not known |
10:52.20 | Follow-me | [2017-01-04 11:51:45] WARNING[30439]: chan_sip.c:5713 create_addr: No such host: start |
10:52.20 | Follow-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) |
10:53.28 | cka | Follow-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) |
11:08.53 | Follow-me | <PROTECTED> |
11:08.53 | 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) |
11:28.06 | *** join/#asterisk follow-me (~Follow-me@212.34.23.94) |
11:28.19 | follow-me | <PROTECTED> |
11:28.19 | follow-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) |
11:37.41 | follow-me | hello |
11:38.11 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
11:41.02 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
11:46.23 | follow-me | cka |
11:46.24 | follow-me | hello |
12:00.05 | defswork | anyone 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) |
12:27.08 | blinky_ | 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) |
12:30.21 | pawiecki | blinky_: other than Wireshark? |
12:30.44 | follow-me | hello |
12:30.51 | pawiecki | follow-me: hi |
12:31.01 | follow-me | anybody here knpw how to do bridged call with originate |
12:31.23 | blinky_ | pawiecki, didn't think of wireshark, will have a look now. |
12:32.19 | pawiecki | blinky_: I mainly use tcpdump to capture packets on linux server, then analyze it in Wireshark. It's very usefull tool. |
12:33.51 | blinky_ | 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. |
12:34.49 | blinky_ | brb got to reboot |
12:39.03 | *** join/#asterisk blinky_ (~damia@host81-136-203-112.in-addr.btopenworld.com) |
12:41.35 | cka | follow-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.. |
12:41.46 | follow-me | <PROTECTED> |
12:50.10 | pawiecki | follow-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) |
13:01.47 | follow-me | I ALREADY did |
13:01.56 | follow-me | and im using amd for outgoing |
13:03.27 | *** join/#asterisk sekil (~sekil@cable-89-216-231-150.dynamic.sbb.rs) |
13:05.08 | follow-me | app_dial.c:2375 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
13:05.16 | follow-me | where do i place subscriber in dial |
13:05.30 | follow-me | Dial("SIP/ipauth2xMJEKBj-00000005", "SIP/000201277682533,,U(default^called_channel^1),30") in new stack |
13:06.18 | pawiecki | follow-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) |
13:23.14 | blinky_ | What is classed as unacceptable jitter? I have 1.359ms at the moment on one call? |
13:26.19 | stefan27 | which function did you use to get data data (1.359 ms)? |
13:26.23 | stefan27 | that data* |
13:26.47 | blinky_ | It was found using tcpdump and looking at the data with wireshark |
13:28.18 | stefan27 | All right, I have a related question on what unit is reported by asterisk's cli command "sip show channelstats" at column "Jitter" |
13:29.40 | blinky_ | bare with me I will check |
13:29.53 | stefan27 | I haven't looked at the mathematical definition(s) of jitter, so I wouldn't know |
13:30.33 | blinky_ | At the moment I have no one in an active call, I will have to wait for someone to make a call |
13:38.57 | stefan27 | I'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) |
13:54.17 | blinky_ | 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) |
13:58.00 | dascodemonkey | I'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? |
14:00.33 | Samot | How do they get to sales now? |
14:06.32 | [TK]D-Fender | dascodemonkey, several different approaches possible for this |
14:07.26 | dascodemonkey | All 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. |
14:08.32 | dascodemonkey | D-fender do you know of any examples I could reference? |
14:10.20 | [TK]D-Fender | dascodemonkey, 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 |
14:10.43 | follow-me | <[TK]D-Fender> welcome back sir |
14:11.27 | follow-me | [zaid] |
14:11.27 | follow-me | exten => start,1,Dial(SIP/${EXTEN},10) |
14:11.31 | yang | Is there some "second hand gear" European VoIP forum, where I could sell VoIP stuff ? |
14:11.35 | [TK]D-Fender | You aren't listeing |
14:11.39 | follow-me | [default] |
14:11.39 | follow-me | <PROTECTED> |
14:11.39 | [TK]D-Fender | I rtold you to use a DIAL OPTION |
14:11.44 | [TK]D-Fender | and you still aren't |
14:11.47 | follow-me | <PROTECTED> |
14:11.49 | follow-me | but 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-Fender | USE THE DIAL OPTION ONLY |
14:12.20 | [TK]D-Fender | NOTHING ELSE |
14:13.20 | follow-me | dial option should be in [default |
14:13.24 | pawiecki | [TK]D-Fender: do you mean GoSub? What's wrong with it? |
14:14.00 | [TK]D-Fender | pawiecki, he should nott be using tthis |
14:14.21 | [TK]D-Fender | I wasted HOURS on this garbage when he can'tt use ONE STUPID DIAL PARAMETER |
14:14.49 | [TK]D-Fender | follow-me, I told you 100 times to DIAL the stupid target and pass a GOSUB or MACRO DIAL OPTION |
14:15.25 | [TK]D-Fender | Read Dial()'s INSTRUCTIONS and put the DIAL OPTION for it |
14:16.22 | [TK]D-Fender | yang, https://www.craigslist.ca/about/sites#EU |
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14:18.41 | yang | [TK]D-Fender: I thought about VoIP-specific forums... |
14:19.20 | follow-me | im afraid im using kolmisoft asterisk |
14:19.22 | follow-me | compiled |
14:19.28 | follow-me | thats why it didnt work i think |
14:19.35 | [TK]D-Fender | yang, In North America we've got Kijiji & Craigslist. Being in a major city there is a huge amount of stuff available |
14:19.38 | follow-me | dial in mor_app |
14:19.56 | [TK]D-Fender | As for "some web forum", no idea. I'm wondering how anyone gets tto finding one so specific to voip. |
14:20.18 | yang | ok |
14:20.50 | yang | [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-Fender | follow-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-Fender | yang, I'm unaware of resale sites & services for EU unfortunately. I suppose there's always ebay... |
14:22.23 | pawiecki | If 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? |
14:22.43 | sekil | hi |
14:23.26 | yang | [TK]D-Fender: There is EBAY, true |
14:24.50 | [TK]D-Fender | pawiecki, pawiecki if A is NAT'd then the packet should be sourced with the NAT address |
14:24.51 | sekil | pawiecki: if you got 200 OK should be ok |
14:25.08 | [TK]D-Fender | sekil, Not if * can't match it to the request |
14:25.21 | Samot | follow-me: MOR != Asterisk. I thought you said you were switching from one to the other? |
14:25.43 | sekil | pawiecki: although I've seen providers sending 200 OK with additional X-headers which say that the money is zero |
14:26.25 | sekil | [TK]D-Fender: I meant he got the 200 OK back to his internal * |
14:26.59 | [TK]D-Fender | sekil, We're gettting implications of it coming back to a different IP that it was sent from |
14:27.37 | sekil | [TK]D-Fender: yeah..nat issue... |
14:28.06 | [TK]D-Fender | but I'm not seeing debug for this |
14:28.09 | [TK]D-Fender | which we SHOULD |
14:28.16 | sekil | pawiecki: I guess if you have static ip you can use externip |
14:28.21 | pawiecki | [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-Fender | because tracking somebody's re-written story gets convoluted for nothing |
14:28.29 | sekil | pawiecki: or you're on dynamic..you can use some stun server |
14:29.20 | [TK]D-Fender | pawiecki, Show the actual debug |
14:29.38 | pawiecki | [TK]D-Fender: ok, please wait a bit. |
14:29.41 | sekil | pawiecki: which IP has changed? |
14:30.37 | sekil | pawiecki: on the router? |
14:31.46 | sekil | hates nat |
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16:40.46 | igcewieling | sekil: weird, it has been years since I've seen a major NAT issue with Asterisk. |
16:42.31 | igcewieling | We generally don't allow our users to be on dynamic IPs, however. |
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20:20.20 | ctjctj | Hello. 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. |
20:21.46 | ctjctj | Think "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" |
20:22.05 | igcewieling | you would do that on the other side of the originate |
20:22.29 | ctjctj | igcewieling, I don't follow. What is the "other side of the originate?" |
20:22.44 | igcewieling | ctjctj: what is your originate command? |
20:23.07 | igcewieling | I can explain in the terms of .call files, which are similar enough it might be applicable. |
20:24.40 | ctjctj | Channel = "SIP/trunk/4435551212", Context="outbound" Exten=4321, CallerID="example.com <201-555-1212>", Variable= (stuff), Priority = 1) |
20:25.16 | igcewieling | you would playback the message in exten => 4321 section, usually using playback and read |
20:25.30 | ctjctj | *nods* Yes. |
20:26.30 | [TK]D-Fender | So HOW do you want to log those? |
20:26.45 | [TK]D-Fender | Do 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-Fender | Direct to file, DB, etc. Or shove our AMI messages for VIR selections, etc |
20:27.10 | [TK]D-Fender | IVR* |
20:27.17 | file | no no, not direct to me |
20:27.35 | ctjctj | [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. |
20:27.58 | ctjctj | The events have Uniqueid, but I don't see how to map it to an ActionID |
20:28.10 | igcewieling | you could collect the digits with READ, then generate a manager or CEL event with the info. |
20:28.14 | [TK]D-Fender | stuff file |
20:28.15 | [TK]D-Fender | :p |
20:28.52 | [TK]D-Fender | ctjctj, When you originate a channel you can a CHANNEL variable to uniquely make tthem identifiable |
20:29.03 | wyoung | happy new year to you [TK]D-Fender |
20:29.06 | [TK]D-Fender | ctjctj, 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 |
20:31.06 | ctjctj | [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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20:55.11 | dmasiero | Anyone know of a device that will stream on-premise hold music back to a hosted Asterisk PBX? |
21:09.41 | Samot | One that will act as a streaming server for Asterisk to be a client for? |
21:10.41 | igcewieling | A HOSTED Asterisk? |
21:11.39 | Samot | I'm guessing this is going to stream over the Internet. |
21:11.51 | dmasiero | At client location is a device that generates music, has a 3.5 mini jack out. Client's PBX is Asterisk in the cloud. |
21:12.15 | Samot | If you can't attach a physical device to the PBX |
21:12.37 | Samot | You'll have to do some sort of streaming client/server setup |
21:13.44 | dmasiero | Was 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. |
21:15.29 | Samot | It wouldn't be a SIP channel. |
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21:21.44 | Samot | You have to setup the musiconhold.conf to call on an application like mpeg123 that can stream from url.. |
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22:36.04 | igcewieling | asterisk'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. |
22:38.40 | igcewieling | Have you looked in musiconhold.conf.sample? You should. |
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23:11.10 | Samot | I think the real issue is the MoH unit is analog and doesn't do streaming. |
23:11.20 | Samot | So they would need a completely new unit or a service this does this. |
23:11.52 | Samot | And hopefully a decent connection. |
23:12.01 | Samot | Phones, Data, MoH streaming... |
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