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| 03:00.34 | ChannelZ | zapotah, in regards to your query last night, I use a super-complex tactic at home. I recorded a prompt that says "If you're a human being, press 3." I then have an extension 3 which rings my phone. |
| 03:00.49 | ChannelZ | I get **0** robocalls. |
| 03:02.06 | ChannelZ | Humans don't have to do Phone Olympics to talk to me with random 3 digit codes. |
| 03:04.29 | ChannelZ | But if you really feel you need to make a keypad game out of it, it's not that hard. Store your sequence in a channel variable. There is a RAND function. Use some prompt and SayDigits() to tell the human what to do. Use Read() to read their reply. React accordingly. |
| 03:06.10 | ChannelZ | But just a hint: If you immediately answer the channel and play something short like "Please press 5 to continue" and wait for 3 seconds for them to do so or hang up on them otherwise, you will probably solve 99% of your problems. |
| 03:07.40 | ChannelZ | Most telemarketer scums are still behind robodialers that do delayed bridging, and will almost never hear the instructions and get hung up on. I see it on my console every day. |
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| 04:07.07 | [[thufir]] | when a softphone sends a password, that's the "secret" entry in sip.conf for that context? |
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| 04:17.28 | [[thufir]] | a 401 unauthorized isn't an error, per se, it's a challenge to a softphone to authenticate? http://pastebin.com/wRi84gBD |
| 04:19.47 | [TK]D-Fender | yes |
| 04:19.54 | [TK]D-Fender | It can mean EITHER |
| 04:20.04 | [TK]D-Fender | Some devices don't like being challenged. |
| 04:21.28 | [TK]D-Fender | in more recent *'s there is an option to "alwaysauthreject" which is "yes" be default to always reject to help keep account names hidden |
| 04:33.28 | [[thufir]] | [TK]D-Fender: I'm using Jitsi and CSipSimple. both get 401 in the log for asterisk, but it's not showing up in the console with "sip set debug on". that seems striking, but I don't know what to make of it. I should look into jitsi logs? I think I have a Jitsi log, it didn't seem informative to me. |
| 04:37.46 | [TK]D-Fender | Show another full comm unfiltered |
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| 05:08.04 | zapotah | ChannelZ: cheers, ill look into that |
| 05:08.53 | zapotah | ChannelZ: its remarkable how difficult it was to find information such as that :) |
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| 05:39.35 | jzu_ | yay, got intenal calls working between two Asterisk's |
| 05:39.57 | jzu_ | IAX2 Trunks and done |
| 05:40.56 | jzu_ | IAX2 should work though NAT without port forwarding, riight? |
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| 06:51.58 | ChannelZ | jzu_, yes and no. They are easier than SIP because everything (including RTP) goes over one port, so firewalls have an easier time with it because of the way they handle UDP |
| 06:56.35 | ChannelZ | You'll still want to use 'qualify' to make sure the firewall(s) don't unmap the port |
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| 07:18.12 | jzu_ | [Helsinki] |
| 07:18.12 | jzu_ | qualify=yes |
| 07:18.12 | jzu_ | qualifyfreqok=25000 |
| 07:19.05 | jzu_ | I did the IAX2 Trunk proof-of-concept in my own LAN, next step is moving the other Asterisk installation to the other site and configuring correct IP's |
| 07:19.25 | jzu_ | but yeah uhm, I do need to enable port-forwarding anyways because both installation sit behind NAT |
| 07:20.01 | jzu_ | these are just two home installations that I try out for fun |
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| 07:43.47 | ChannelZ | Well that's why I say "yes and no". The easy way is yes, just do it. But since UDP doesn't have state, the way NAT generally works is that when you send a UDP packet to someone on port XXX, the firewall maps incoming port XXX back to that machine for replies. |
| 07:45.57 | ChannelZ | Since IAX only uses only one port number that's the same, if both sides register with each other and/or otherwise keep sending packets to each other often enough to keep the NAT table active, it technically should work without forwarding. |
| 07:46.28 | ChannelZ | But if you can port forward, just do it and be done with it. |
| 07:47.35 | jzu_ | yeah, I can and will - it will be "safer" in the end |
| 07:48.12 | jzu_ | wonder what I should do so I could get Sonetel to forward their number to my Asterisk at home, mmmm |
| 07:48.41 | jzu_ | http://www.sonetel.com/site/pmwiki.php?n=Admin.ConnectingAsterisk - I mean, do I need port forwards for that since my Asterisk is behind NAT? |
| 07:48.58 | jzu_ | "yes and no" is my guess :-) |
| 07:49.55 | jzu_ | idea is that I can make internal calls between two Asterisks that are connected through IAX2 Trunk and both would have public numbers from Sonetel for outgoing calls |
| 07:50.36 | jzu_ | "nat=yes", so I'll just test it - it's not rocket science :-) |
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| 11:29.34 | oli-werk | i'm trying to create a dialplan that: |
| 11:30.32 | oli-werk | when a certain extension is called, it dials an external number and plays a message to the person that answers |
| 11:30.48 | oli-werk | then after playing that message, connects the caller and callee together |
| 11:31.22 | oli-werk | with Dial(blah,balh,G(default^blah^1)) |
| 11:31.43 | oli-werk | but i'm stuck with a 403 forbidden error when it reaches the dial |
| 11:32.25 | oli-werk | [2015-03-25 22:27:45] VERBOSE[5114][C-0000613a] app_dial.c: -- Called SIP/onehundredexe/01010100110 |
| 11:32.25 | oli-werk | [2015-03-25 22:27:45] WARNING[17107][C-0000613a] chan_sip.c: Received response: "Forbidden" from '"ext name" <sip:7122@123.123.123.123>;tag=as666f732c' |
| 11:32.33 | oli-werk | any ideas? |
| 11:34.50 | oli-werk | [speak-test] |
| 11:34.51 | oli-werk | exten => 5555,1,NoOp() |
| 11:34.51 | oli-werk | <PROTECTED> |
| 11:34.51 | oli-werk | <PROTECTED> |
| 11:35.21 | oli-werk | ... followed by some playback and bridging logic |
| 11:37.14 | WIMPy | Dial can play a message to the called party directly. |
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| 11:39.31 | oli-werk | with ,A yup |
| 11:39.45 | oli-werk | still a WIP, but need to work out how to get it dialling out first |
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| 11:57.24 | oli-werk | looks like it tries to call directly out the trunk rather than doing an invite |
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| 12:00.45 | oli-werk | or rather, it tries to invite the dialed number |
| 12:01.21 | oli-werk | but the from is <sip:extension@pbx> to <sip:number@trunk> |
| 12:01.50 | WIMPy | What the heck are you on about? |
| 12:02.15 | oli-werk | 403 forbidden when an extension tries to call out |
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| 12:02.22 | oli-werk | sip debug show the above |
| 12:02.28 | WIMPy | Dial places a call somewhere. And with A you can make it play an announcement befor it connects the two parties. |
| 12:02.47 | WIMPy | So that's no longer related to the other story? |
| 12:02.55 | oli-werk | my question was |
| 12:02.59 | oli-werk | <oli-werk> but i'm stuck with a 403 forbidden error when it reaches the dial |
| 12:04.19 | oli-werk | the sip invite appears to come from the extension@pbx to number@trunkhost |
| 12:04.39 | oli-werk | instead of username/cid@pbx to number@trunkhost |
| 12:04.42 | oli-werk | not sure why |
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| 12:13.53 | Onyx47 | Hello all. Anyone familiar with executing AGI scripts from bash? I'm getting a broken pipe when I try to execute SendDTMF. Pointing me towards an example would help too, all I keep finding is phpagi, almost no bash examples around |
| 12:14.58 | [TK]D-Fender | You don't execute those from bash |
| 12:15.10 | [TK]D-Fender | You can have a bash script AS an AGI though.... |
| 12:15.26 | [TK]D-Fender | ~book |
| 12:15.26 | 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 |
| 12:15.28 | [TK]D-Fender | ^^^^ |
| 12:16.17 | [TK]D-Fender | AGI is well documented in there. Also clearly no-one really cares to write AGI's in bash much so they didn't bother to make any libraries for it so you just have to do it all yourself as per the docs |
| 12:21.17 | Onyx47 | [TK]D-Fender, I wouldn't bother with bash either usually, but I literally need to execute a single command for one specific client and I don't use AGI anywhere else, doing it in PHP or something seems like an overkill. Thanks for the link. |
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| 12:22.14 | WIMPy | Onyx47: The key to AGI is that you can't just send somethign back without reading what you get. |
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| 12:23.14 | Onyx47 | WIMPy, yeah, I'm executing a read. I have a hunch that it's timing out though and I'm trying to find out what the proper way to do it in bash is. |
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| 12:24.07 | [TK]D-Fender | Onyx47: sHOW US WHAT YOU'RE ACTUALLY DOING. |
| 12:24.20 | [TK]D-Fender | </oopscaps> |
| 12:26.11 | Onyx47 | [TK]D-Fender, http://pastebin.com/Yewcjz4D |
| 12:26.17 | Onyx47 | that is honestly all I need |
| 12:26.48 | [TK]D-Fender | Onyx47: you have to do the FIRST read..... |
| 12:26.48 | Onyx47 | I'd do it in a macro / sub but BRIDGEDPEER doesn't get set yet, and I can't send DTMF on the caller channel for that reason |
| 12:26.52 | WIMPy | You need to read before you send. |
| 12:27.01 | [TK]D-Fender | Onyx47: AGI's are GIVEN data at the start that you have to read in.... |
| 12:27.16 | [TK]D-Fender | Onyx47: And you aren't using the right interfaces for that... |
| 12:28.03 | Onyx47 | [TK]D-Fender, I never messed with AGI before, can you explain what you mean by "right interfaces"? |
| 12:28.16 | [TK]D-Fender | Onyx47: Read the chapter on AGI...... |
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| 12:31.02 | Onyx47 | Getting on it now. I'd do this in AMI to be perfectly honest, but since the dialplan is self-contained in every other case I'd really like to avoid depending on an external service for this. |
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| 13:18.42 | Onyx47 | So, as it turns out even the example from the book, which looks completely reasonable to me, throws the same errors. It works, it just keeps throwing errors. |
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| 13:20.10 | budman_mtl | good morning room |
| 13:22.40 | [TK]D-Fender | Onyx47: Show, don't tell. |
| 13:27.08 | Onyx47 | [TK]D-Fender, http://pastebin.com/LEsTkx69 |
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| 13:30.11 | [TK]D-Fender | enable AGI debug and retest |
| 13:30.21 | [TK]D-Fender | agi <tab> till you get the syntax |
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| 13:32.00 | [TK]D-Fender | Actaully... |
| 13:32.02 | Onyx47 | It's on. All I got is in the pastebin. |
| 13:32.07 | [TK]D-Fender | I do see it there... |
| 13:32.10 | [TK]D-Fender | yeah |
| 13:32.21 | Onyx47 | it's more annoying than anything at this point |
| 13:32.21 | [TK]D-Fender | ok, I see it issuing the commands... then erroring only at the end |
| 13:32.35 | Onyx47 | yeah, it looks like it can't write to STDOUT |
| 13:32.40 | Onyx47 | which is kinda insane |
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| 13:34.36 | [TK]D-Fender | [Mar 25 14:13:02] <SIP/609R0001-00000073>AGI Rx << SAY ALPHA "Hello World" "" |
| 13:34.39 | [TK]D-Fender | It's getting it... |
| 13:34.46 | Onyx47 | yeah, it's playing |
| 13:35.06 | [TK]D-Fender | You might need an extra read in there... usually that error means that there is still more unread before you can write |
| 13:35.16 | [TK]D-Fender | loop it like you do at the start |
| 13:36.33 | Onyx47 | well, in my case the only thing I can really do with the response is retry if I get a failure which... I mean it's SendDTMF, the only way I can see it failing is the channel getting destroyed, and at that point I don't care anyway, but still, annoying |
| 13:36.50 | [TK]D-Fender | So loop the read like you do at the start |
| 13:36.51 | Onyx47 | I tried that. I'll give it another go though, one minute |
| 13:37.43 | Onyx47 | yeah, identical results |
| 13:38.35 | Onyx47 | hmmm... I wonder how it terminates it. Maybe read detects a terminator too early and closes the pipe prematurely. |
| 13:41.56 | [TK]D-Fender | might have to dig just a little... after maybe living with it for now if it survives enough to do what you need... |
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| 13:47.47 | Onyx47 | Oh well, as long as it works.. I suspected that the space character might be confusing read, tried setting IFS variable to newline only, no change |
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| 13:49.27 | oli-werk | i'm trying to create an extension which: calls an outside number, plays a message to the called party and gives them an option to hangup by pressing 0 or continue by pressing 1 |
| 13:49.43 | oli-werk | and continue would connect them with the caller that dialed the extension |
| 13:50.19 | oli-werk | i've tried using Dial() pointing to a macro, but that doesn't seem suitable for ivr like options |
| 13:50.34 | oli-werk | and also tried dial() G splitting the channel |
| 13:50.41 | oli-werk | and then bridging them back together... no luck |
| 13:50.44 | oli-werk | any suggestions? |
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| 13:55.58 | Onyx47 | oli-werk, well... I don't know if it's the best solution and it's a bit convoluted, but you could hack it together using a callfile and a Local extension |
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| 14:04.11 | [TK]D-Fender | oli-werk: "core show application dial" <- Macro or Gosub. Take your pick. |
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| 14:16.56 | eirirs | <PROTECTED> |
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| 14:23.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:35.29 | oli-werk | thx [TK]D-Fender |
| 15:35.32 | oli-werk | got there in the end |
| 15:58.28 | marceloamorim | Guys, I don't know if this is a good question, but in a transfer using the atxfer, there is a possibility to cancel that transfer ? |
| 15:59.41 | marceloamorim | like, I receive a call, I'll transfer that call with the featuremap > atxfer, when I transfer to an extension I realized that is the wrong extension, I want to cancel that transfer.. is it possible? |
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| 16:09.01 | ChannelZ | I think if the called party hangs up, you should be reconnected to the transferee. Other than that I dunno.. like if you wind up in someone's voicemail.. |
| 16:11.35 | marceloamorim | actually ChannelZ, I'm reconnecting the the transferee, but I want to cancel by my own |
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| 16:35.53 | kb3ien | What do we like for H264 with asterisk (on android) ? |
| 16:36.01 | cervajs2 | somebody using queue + local channels there? |
| 16:36.10 | [TK]D-Fender | Where is "there"? |
| 16:36.40 | cervajs2 | as member in queue |
| 16:36.58 | cervajs2 | member => Local/88@route_phones_1,3,angelina_jolie,SIP/virtual_88 |
| 16:37.27 | [TK]D-Fender | Ask your actual question. Weasels also eat people who ask meta-questions.... |
| 16:38.02 | cervajs2 | i'm facing problem with multiple calls to one agent when Local channels are used |
| 16:39.11 | cervajs2 | wireshark shows multiple invites to the agent's phone |
| 16:39.23 | [TK]D-Fender | You never need to use wireshark for this |
| 16:39.34 | [TK]D-Fender | * tells you what it's dialing from CLI |
| 16:39.40 | cervajs2 | queue is configured with ringinuse=no, autofill=yes |
| 16:39.55 | [TK]D-Fender | You've clearly done something wrong... so let's looks at all the relevent configs |
| 16:40.03 | [TK]D-Fender | And status dumps |
| 16:40.28 | [TK]D-Fender | Actually I see one thing I suspect right away.... |
| 16:40.36 | [TK]D-Fender | But lets see the rest |
| 16:41.52 | cervajs2 | ok. what do you want? |
| 16:43.54 | cervajs2 | i can simulate this situation with sipp. i see that call goes to agent and before he answer it, asterisk sends another call to the same agent |
| 16:45.03 | cervajs2 | it looks like there is some problem with device state. tested on ast1.8 and ast13 |
| 16:47.57 | jzu_ | hmmm |
| 16:48.08 | jzu_ | W320 1 Error(s) Parsing: SIP003094C3139C.cnf <-- any clue how to debug that using telnet/ssh to the phone?-) |
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| 16:54.38 | Section1 | hi guys question...we record all incoming calls and i want that in the filename show the sip user who take that call...its possible ? |
| 16:55.38 | Section1 | i have this line now: exten => s,n,MixMonitor(/var/spool/asterisk/monitor/call/call-${STRFTIME(${EPOCH},,%Y-%m-%d--%H-%M-%S.wav)},b) |
| 16:57.02 | Section1 | i want something similar to call-EXT-SIPNUMBER-date-time.wav |
| 16:57.42 | Section1 | and EXT-SIPNUMBER is the person who take that incoming call |
| 17:06.05 | marceloamorim | ChannelZ: hey, I thought the disconnect on the featuresmap was for disconnect the first call and you can use for cancel the transfer |
| 17:06.39 | [TK]D-Fender | Section1: "core show application dial" <- |
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| 17:07.13 | [TK]D-Fender | Section1: You can start your recording AS they bridge, or set an "actual" filename to save as so you can rename it at the end of the call, etc |
| 17:09.48 | Section1 | [TK]D-Fender, i get "Not available" |
| 17:10.05 | [TK]D-Fender | What "not available"? |
| 17:10.52 | Section1 | that command ...like synopsis Not available |
| 17:11.47 | Section1 | but no problem i will try some stuff and checj |
| 17:11.50 | Section1 | check* |
| 17:11.53 | Section1 | thanks |
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| 17:42.28 | rrittgarn | Question on app_voicemail: After a caller leaves a message and hits # or hangs up, does the system write the file out, and then attach that file to the email listed in voicemail.conf, or does the system send the email and then write it to disk? |
| 17:43.19 | rrittgarn | having a user claim they never dialed in for a voicemail, received the email but state it wasn't there. I find in the logs where it looks like a message of the same number was deleted within the expected time frame, but cannot be 100% certain on the message contents as I do not record internal calls. |
| 17:43.50 | rrittgarn | So if the order of operations is Write to disk -> Email, I feel confident in that someone there deleted the message |
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| 21:00.11 | kplong | Hello. General question, does Asterisk support video calling on any major brand (ie: Cisco) phones that have video screens? I'm running FreeBPX currently and have been asked to set up video calling. |
| 21:02.01 | WIMPy | yes |
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| 21:15.22 | epaphus | Hello all. |
| 21:15.35 | epaphus | Could anybody kindly explain to me this short dial plan? What does it match? (**xxx.|[69]xxxxxxxxS0|10xxxxS0|1xxxS0|00xxxxxx.) |
| 21:16.01 | WIMPy | Check the manual of the device it belongs to. |
| 21:16.18 | epaphus | its for linksys |
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| 21:48.51 | kplong | WIMPy: great. Have you done it before? Looking for a phone recommendation, preferably Cisco |
| 21:53.28 | WIMPy | No |
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| 22:06.32 | phix | WIMPY!!!! |
| 22:16.15 | pjensen00 | I have two SPA942s on my desk right now |
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| 22:16.25 | pjensen00 | Sound quality is lacking on them. |
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| 22:16.33 | pjensen00 | But useful for testing. :P |
| 22:19.09 | Penguin | epaphus: x should match numbers 0-9 |
| 22:19.20 | Penguin | epaphus: . should match more characters/digits |
| 22:19.51 | Penguin | epaphus: I think S0 means dial immediately when the number of digits defined is reached. |
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| 22:33.41 | tdonahue | Hi all, I have a bunch of yealink T46 phones behind a firewall that are registering to a hosted asterisk server. I keep getting "Unregistered SIP ..." followed by "Registered SIP..." messages in the console |
| 22:34.36 | tdonahue | has anyone else seen the Yealink phones doing this? These are new phones with the 73.x firmware installed on them, and I don't remember the 72.x firmware behaving the same way |
| 22:41.25 | Penguin | It makes me think of NAT closing off. |
| 22:45.40 | tdonahue | its weird, when I have seen the logs where it was the firewall closing the port the peer always goes to unreachable... the logs actually show the phone un-registering then re-registering a couple seconds later |
| 22:46.56 | tdonahue | on a lark, i turned off STUN on the account and it seems to have stopped unregistering... maybe it has something to do with how yealink was handling STUN |
| 22:47.25 | Penguin | If asterisk is configured right and if the NAT plays nicely, you don't need STUN anyway. |
| 22:47.53 | Penguin | But they probably don't have a NAT on the asterisk side anyway. |
| 22:48.19 | tdonahue | yeah, i've been using asterisk for so long (since the 0.9 days), its force of habit |
| 22:49.13 | tdonahue | i configure it on the phones without realizing that i'm doing it half of the time |
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| 23:47.16 | dlynes | Is there a way to index arg1/arg2/arg3/... programmatically? ${ARG${COUNT}} doesn't seem to work |
| 23:48.19 | Penguin | Can you CUT it first? |
| 23:49.09 | dlynes | Penguin: like say if i want to loop through all the parameters of a macro, treating each parameter the same |
| 23:51.15 | dlynes | Or perhaps a way to take a string and convert it to a variable name |
| 23:51.44 | Penguin | Example? |
| 23:51.53 | [TK]D-Fender | EVAL <- |