| 00:04.16 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
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| 00:36.29 | c|oneman | is there a voip/sip troubleshooting channel? I have an aastra phone |
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| 01:28.09 | fling | Are g723 and g726 good? |
| 01:29.11 | *** join/#asterisk mintos (~mvaliyav@114.143.44.1) |
| 01:31.57 | hebber | Never tried them |
| 01:39.57 | fling | I have a lot of these warnings in console > WARNING[9773]: chan_sip.c:7370 in sip_write: Can't send 10 type frames with SIP write |
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| 02:41.55 | igcewieling | Asterisk does not fully support g723.1 so don't use it |
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| 03:57.51 | *** join/#asterisk ledoktre (~ledoktre@bb-155-182.omnitelcom.com) |
| 03:59.43 | ledoktre | Greetings all. I was on here earlier working with some fellas trying to figure out a regex that would set my ${CALLERID{NUM}) to UNKNOWN if the variable presently contained anything but numerals. I had some issues with the proper usage of {} [] and () (when to use them), but I think Ive got that sorted out. Right now, with a simple test, I cannot get the regex to parse : |
| 04:01.55 | ledoktre | Set(test=${REGEX("[\D]" 1AA2345)}) and Set(test=${REGEX("[\D]" 12345)}) with either one, it returns test=0 |
| 04:02.37 | ledoktre | I would have expected the first one to return 1 as it contains "AA" in the string? |
| 04:04.10 | igcewieling | have you tried [^\d] maybe Asterisk's regex does not support \D |
| 04:05.06 | ledoktre | igcewieling : so we meet again ;) No, I haven't tried that one. Im pretty new to regex |
| 04:05.10 | ledoktre | I'll give it a shot |
| 04:05.13 | igcewieling | or adding a + after the ] |
| 04:06.36 | igcewieling | or better yet just to be paranoid "^[^\d]+$" |
| 04:07.32 | igcewieling | BTW, most people put the num and name in lower case. CALLERID(num) because that is how it is documented |
| 04:10.25 | ledoktre | igcewieling: thanks for the tips regex and syntax. Thats all I was after earlier, I didn't intend for it to turn into anything. |
| 04:11.34 | apb1963 | watches all the hot groupies groping igcewieling |
| 04:13.34 | apb1963 | Are those screams of terror I hear? |
| 04:14.02 | ledoktre | igcewieling: *sighs*. I I've got two lines s,2,Set(testnum=${REGEX("^[^\d]+$" 12345)}) and s,3,Set(testalpha=${REGEX("^[^\d]+$" 12AA345)}). The output is testnum=1 and testalpha=1. I think I'm either missing something obvious or it just hates me ;) |
| 04:15.08 | igcewieling | regex is not easy. ask on #regex |
| 04:15.18 | igcewieling | and go read the book |
| 04:15.33 | igcewieling | using numbered priorities is like driving a model-t |
| 04:16.00 | apb1963 | those are worth a lot of money regardless of condition |
| 04:16.07 | apb1963 | :) |
| 04:16.15 | igcewieling | ~book |
| 04:16.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:18.55 | ledoktre | igcewieling: I'll have to read up on what you mean by not using numbered priorities (but another day I guess). Ive used Asterisk for a number of years but never really got into it. Never knew anything else came down the pipe besides numered priorities ;) |
| 04:19.02 | ledoktre | thanks though. |
| 04:19.46 | igcewieling | ledoktre: The "n" priority was added around 2005 |
| 04:22.12 | ledoktre | igcewieling: time flies I guess when your having fun |
| 04:22.33 | apb1963 | ledoktre: Replace your priority numbers with an "n", except for the first one ("1"), and you're good to go. |
| 04:23.26 | ledoktre | apb1963: thanks, I'll give that a try. The last time I got into trying to program stuff was back when I bought this yellow book, "VoIP Telephony with Asterisk" by Paul Mahler. I bought it back when it was new (2003? 2004?) |
| 04:23.44 | ledoktre | Still got it on the shelf (though it probably doesn't cover much anymore) |
| 04:24.37 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.113) |
| 04:30.04 | apb1963 | does asterisk support the full regex library? Or does it have it's own internal idea of what regex should do? |
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| 04:39.08 | ledoktre | igcewieling: In case you are curious, I figured it out (mostly). You can see the details here : https://issues.asterisk.org/jira/browse/ASTERISK-20715 |
| 04:39.08 | LieutPants | [ASTERISK-20715] [Status: Closed] REGEX function ignores shorthand character starting with backslash - https://issues.asterisk.org/jira/browse/ASTERISK-20715 |
| 05:01.07 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 05:03.47 | linocisco | hi all |
| 05:04.01 | linocisco | is there any digium asterisk training in Thailand? |
| 05:06.31 | fling | Is 'Panasonic KX-TG6611' good if I need a cheap dect phone? |
| 05:07.03 | fling | I have an old d-link pata, I want to use CallerID |
| 05:11.15 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 05:45.54 | linocisco | fling, check this out http://www.google.com/url?q=http%3A%2F%2Fwww.citel.com%2FProducts%2FPortico%2FSpec_Sheets%2FPresentation.pps |
| 05:46.00 | linocisco | save ur money |
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| 05:48.52 | linocisco | http://www.citel.com/Products/Portico/Spec_Sheets/Presentation.pps |
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| 06:07.38 | fling | linocisco: Presentation.pps: Composite Document File V2 Document, corrupt: Can't read SAT |
| 06:08.23 | fling | What is that? |
| 06:10.04 | kaldemar | nothing useful. |
| 06:10.34 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:11.13 | kaldemar | just an ATA, called a TVA this time. |
| 06:12.54 | fling | kaldemar: hello ;p |
| 06:19.23 | kaldemar | howdy |
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| 07:37.44 | goddva | We have a small server where we have PostgreSQL 9.2.4 installed together with Asterisk (VOIP) service. We used to run the same setup with MySQL, but had major problems with locking, so therefore the change. Now we are experiencing choppy/stuttering sound during calls, and we really cant figure out why. The memory is low, the IO is low, and the CPU is low. Any ides where to start? |
| 07:41.07 | ectospasm | what technology (DAHDI/SIP/etc...) are these calls using? |
| 07:44.49 | ectospasm | goddva: ^ |
| 07:50.25 | goddva | SIP only |
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| 07:50.56 | goddva | no transcoding, alaw to alaw |
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| 08:20.26 | bulkorok | hi |
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| 08:52.09 | mirela666 | Hi all |
| 08:52.42 | mirela666 | <PROTECTED> |
| 08:53.11 | mirela666 | odbc read ODBC_AgentByCode HOT,48 exec |
| 08:53.12 | mirela666 | interface SIP/FonliderPBX/381648854461 |
| 08:53.12 | mirela666 | Returned 1 row. Query executed on handle 0 [LocalMySQLD] |
| 08:53.50 | mirela666 | But when I try to retrive it, value is empty |
| 08:53.53 | mirela666 | Set(ResultSet=${ODBC_AgentByCode(${DefaultQueue},${Option})}); |
| 08:54.14 | mirela666 | -- Executing [~~s~~@macro-DivertWithCode:1] Set("SIP/mirko-0000000a", "ResultSet=") in new stack |
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| 09:05.17 | *** join/#asterisk eject_ck (~Eugene@95.67.72.22) |
| 09:05.33 | eject_ck | hi |
| 09:05.35 | eject_ck | [2013-06-07 12:05:02] WARNING[2703]: db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database |
| 09:05.45 | eject_ck | What DB this is about ? |
| 09:06.09 | kaldemar | astdb, i.e. the internal database. |
| 09:10.26 | kaldemar | mirela666: what do you have in ${DefaultQueue} and ${Option}? |
| 09:11.40 | mirela666 | kaldemar: HOT,48 strings for examle |
| 09:12.23 | kaldemar | that's what you assume them to contain. what do they contain? |
| 09:12.29 | mirela666 | kaldemar: for that exact example |
| 09:13.08 | mirela666 | I logged them with Log and got thesame thing |
| 09:13.11 | mirela666 | a sec |
| 09:14.14 | mirela666 | kaldemar: LOG(ERROR,DefaultQueue ${DefaultQueue}); |
| 09:14.14 | mirela666 | <PROTECTED> |
| 09:14.14 | mirela666 | <PROTECTED> |
| 09:15.16 | mirela666 | <PROTECTED> |
| 09:15.17 | mirela666 | [Jun 7 11:14:44] ERROR[22236]: Ext. ~~s~~:1 @ macro-DivertWithCode: DefaultQueue HOT_PERSONAL |
| 09:15.17 | mirela666 | <PROTECTED> |
| 09:15.17 | mirela666 | [Jun 7 11:14:44] ERROR[22236]: Ext. ~~s~~:2 @ macro-DivertWithCode: Option 48 |
| 09:15.17 | mirela666 | <PROTECTED> |
| 09:15.20 | Greenlight | ~pb |
| 09:15.21 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 09:15.54 | mirela666 | http://pastebin.com/1XJFabA9 |
| 09:16.06 | mirela666 | sorry |
| 09:17.12 | mirela666 | aha they are different |
| 09:17.20 | kaldemar | the output doesn't match your CLI command. |
| 09:19.56 | mirela666 | yes, I use like in sql but in a wrong way |
| 09:20.15 | Greenlight | If I comment out a property against a peer in sip.conf, and "sip reload" doesn't pick up the change, would that be considered a bug? |
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| 09:29.54 | mirela666 | kaldemar: Thx for felp, Got it working now :) |
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| 11:16.48 | liamjfoy | I have everything working in asterisks. However, when I call out the mobile phone receives the call. Upon answer my SIP phone keeps ringing and the mobile user can't hear anything. Is this something to do with RTP? |
| 11:19.43 | kaldemar | liamjfoy: sounds like your asterisk is behind a NAT and you're using a SIP provider to dial out. correct? |
| 11:20.21 | liamjfoy | The asterisk machine is a routable IP. The SIP phone is at home yes (can receive calls). |
| 11:20.53 | liamjfoy | I am using a provider, yes. |
| 11:24.07 | kaldemar | so a NAT between asterisk and the phone? |
| 11:24.15 | liamjfoy | Yes.. |
| 11:25.01 | kaldemar | yo most likely have a re-invite going on and need to disable that. what version of asterisk are you using? |
| 11:26.45 | liamjfoy | asterisk-1.8.20 |
| 11:26.52 | liamjfoy | OK, I'll look into that. |
| 11:27.31 | liamjfoy | Ah, that's directmedia? I have that disabled already |
| 11:28.35 | kaldemar | nat=yes and directmedia=no for the phone. if that doesn't help, enable sip debug and verbosity, make a call and pastebin the output. |
| 11:29.40 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 11:29.48 | StaRetji | Howdy folks. |
| 11:31.13 | StaRetji | I have problem where some times sip user can't hear ringtone. Is there a way to play fake ring if real ringtone is not heard. I was reading progressinband, 180 and 183, but couldn't quite catch it |
| 11:32.13 | *** join/#asterisk Rumbles (~Rumbles@host81-133-228-3.in-addr.btopenworld.com) |
| 11:32.29 | StaRetji | I have option in Gateway SIP 183, but I also have option SIP 180 then 183 |
| 11:32.49 | StaRetji | is SIP 180 then 183 what I am looking for? |
| 11:43.33 | sekil | 180 would mean the far endpoint to do local ringing.. |
| 11:46.43 | StaRetji | thx sekil, I figure that out, but I am not sure does 180 then 183 means to do local ringing until 183? |
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| 12:03.41 | Greenlight | It's unusual to get both 180 and 183 in my experiance |
| 12:04.11 | Greenlight | 183 is session progress, and is mostly the remote end playing ringing |
| 12:04.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:04.41 | Greenlight | Although some IVR's use this to play non-ringing |
| 12:04.48 | Greenlight | So you want to play what it sends to avoid confusion |
| 12:05.01 | StaRetji | thx Greenlight |
| 12:05.31 | StaRetji | I have tested and I alway here telco ringing, which is good |
| 12:05.39 | StaRetji | hear telco ringing* |
| 12:05.46 | StaRetji | but, I have a sip peer |
| 12:05.57 | StaRetji | and he also get's 183 |
| 12:06.04 | StaRetji | but his client doesn't |
| 12:06.09 | StaRetji | :/ |
| 12:06.14 | Greenlight | What do you mean his client doesnt ? |
| 12:06.35 | StaRetji | his sip user, of my sip peer (another sip server) |
| 12:06.53 | StaRetji | hear sound, call is okay, only he doesnt hear ring tone |
| 12:06.54 | Greenlight | So, you send 183 "progress" ? |
| 12:07.09 | StaRetji | yes, because works for me and works for my sip peer |
| 12:07.17 | StaRetji | it doesn't work for his client |
| 12:07.20 | Greenlight | BUt he doesn't hear anything |
| 12:07.37 | StaRetji | it is like this, his client makes a call, no sound until ANSWER |
| 12:07.43 | Greenlight | At a guess I'd think that was a possible NAT type issue |
| 12:07.47 | StaRetji | once ANSWER, he hear sounds |
| 12:07.50 | Greenlight | If you are using directmedia? |
| 12:07.57 | StaRetji | yes, I am |
| 12:08.01 | Greenlight | Yes, thats the issue |
| 12:08.03 | Greenlight | So; |
| 12:08.07 | StaRetji | I should turn it off? |
| 12:08.14 | Greenlight | *before* the lines connects there is no directmedia |
| 12:08.28 | StaRetji | aha |
| 12:08.31 | Greenlight | *after* the line connects you send a RE-INVITE and so end to end media flows |
| 12:08.37 | StaRetji | got it |
| 12:08.44 | Greenlight | The problem is that for some reason the media from your server to him isn't getting through |
| 12:08.59 | Greenlight | Once it's connected, the end points speak direct and so it's a non-issue |
| 12:09.23 | StaRetji | so, actually, if I understand well, directmedia=yes is causing this? |
| 12:09.29 | StaRetji | or I am wrong? |
| 12:09.31 | Greenlight | Umm no |
| 12:09.41 | Greenlight | Well, if you disabled directmedia you'd have no audio whatsoever :)( |
| 12:09.47 | StaRetji | aha, got you |
| 12:09.58 | StaRetji | so, directmedia is helping me to get audio at all |
| 12:10.02 | Greenlight | Double check your network settings in sip.conf |
| 12:10.05 | StaRetji | but is not active until ANSWER |
| 12:10.19 | StaRetji | nat=yes |
| 12:10.26 | StaRetji | but actually, I have public IP |
| 12:10.28 | Greenlight | Yes, directmedia kicks in at answer, when it'll send a "REINVITE" connecting both endpoints RTP |
| 12:10.39 | Greenlight | localnet set correctly ? |
| 12:10.49 | Greenlight | externip set ? |
| 12:10.59 | StaRetji | hm, now that you say, it is not |
| 12:11.15 | StaRetji | should be public ip/255.255.255.0 ? |
| 12:11.35 | Greenlight | externip should be your externalip address |
| 12:11.48 | Greenlight | 255.255.255 isn't an address, it's a netmask |
| 12:11.57 | StaRetji | yes, in my case it is already public |
| 12:12.01 | StaRetji | like 1.2.3.4 |
| 12:12.17 | StaRetji | then, localnet should be 1.2.3.0/255.255.255.0 |
| 12:12.18 | StaRetji | right? |
| 12:12.51 | Greenlight | Usually something line 192.168.1.0/255.255.255.0 |
| 12:13.00 | Greenlight | BUt that depends on your LAN size |
| 12:13.08 | StaRetji | yes, but I dont have local subnet |
| 12:13.15 | Greenlight | Then you don't need that |
| 12:13.16 | StaRetji | I have directly 1 ip on eth0 |
| 12:13.31 | Greenlight | Since there are no local networks |
| 12:13.32 | StaRetji | I can ; comment it? |
| 12:13.42 | Greenlight | Yes, you should be able to |
| 12:13.45 | StaRetji | well, there is VPN at local address |
| 12:13.52 | StaRetji | sorry for confusion |
| 12:13.59 | Greenlight | Is you peer connecting from the internet ? |
| 12:14.05 | StaRetji | yes |
| 12:14.19 | StaRetji | and I have them several, there issue with only 1 |
| 12:14.21 | Greenlight | Just ensure externip is set correctly then |
| 12:14.35 | Greenlight | Hmm... the problem could be at his side then |
| 12:14.55 | StaRetji | how about localnet? I have vpn and gateways connecting trough 10.0.1.0/24 vpn network |
| 12:14.56 | Greenlight | Do you have nat=yes against his peer ? |
| 12:15.02 | StaRetji | yes, I have |
| 12:15.14 | Greenlight | Yes, if you have VPN then treat that as the localnet |
| 12:15.22 | StaRetji | ok, I see now |
| 12:15.27 | StaRetji | directmedia=nonat |
| 12:15.36 | StaRetji | that is my settings in sip.conf |
| 12:16.20 | StaRetji | so, basicaly, I haven't touch anything now, as it is okay, public ip is external ip |
| 12:16.30 | StaRetji | localnet is vpn subnet |
| 12:16.42 | StaRetji | only I am not sure if directmedia=nonat is good or no |
| 12:17.07 | Greenlight | I would tend to have directmedia=no |
| 12:17.21 | Greenlight | But in your case I think this may cause issues with this particular peer |
| 12:17.32 | Greenlight | I suspect they've got a "SIP=-aware" router or something |
| 12:17.37 | Greenlight | At a guess |
| 12:17.48 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:18.03 | Greenlight | To be sure you'll need to get a SIP trace of the call |
| 12:18.10 | StaRetji | I understand |
| 12:18.35 | StaRetji | thank you so much Greenlight, I really appreciate your time |
| 12:18.38 | Greenlight | Right. Lunch :) |
| 12:18.40 | Greenlight | np |
| 12:21.57 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 12:38.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:42.41 | Greenlight | StaRetji: Just had a thought, as a quick fix/hack you *may* be able to get that peer working by doing directrtpsetup=yes |
| 12:45.25 | *** join/#asterisk Tarso (~Tarso@186.215.70.48) |
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| 12:51.11 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 12:55.57 | StaRetji | Hi Greenlight, thx, will add it |
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| 12:58.35 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.238.228) |
| 13:01.23 | StaRetji | Greenlight, 1 more question mate |
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| 13:01.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:01.36 | StaRetji | is it possible to send fake ring only to 1 sip user? |
| 13:01.53 | StaRetji | for example, if I have 3 users, 2 will have early media, no fake ring |
| 13:02.00 | StaRetji | and this one will have fake ring? |
| 13:02.22 | *** join/#asterisk davlefouAMD (~david@41.225.172.115) |
| 13:02.58 | Greenlight | Hmm... the "Ringing" application will send a 180 ringing. However, I'm not sure if there's a way to supress a 183 if that's coming from the other side |
| 13:03.05 | Greenlight | For a single peer |
| 13:03.27 | Greenlight | maybe with an arguemnt to Dial |
| 13:04.00 | Greenlight | Ahh yes, "r" |
| 13:04.30 | Greenlight | So, when that peer makes a call, you'll need to pass the "r" argument to Dial |
| 13:04.49 | Greenlight | However, it's generally bad practice |
| 13:04.52 | StaRetji | I follow you |
| 13:04.59 | Greenlight | Take for example, you're calling into PSTN |
| 13:05.01 | StaRetji | yes, I have r removed for all outgoung calls |
| 13:05.14 | Greenlight | And 183 audio is "This number is currently out of service, please try later" |
| 13:05.24 | Greenlight | You are replacing that with a ringing sound |
| 13:05.28 | Greenlight | So your users keep calling |
| 13:05.43 | StaRetji | yes, yes, well |
| 13:06.26 | StaRetji | still, I have no clue how to pass r to one particular user :/ |
| 13:06.39 | WIMPy | It's not easy to get such announcements to the phones anyway. |
| 13:07.09 | Greenlight | Under normal circumstances such audio would pass though |
| 13:08.15 | WIMPy | What do you call normal? I had lots of issues trying to get that with dahdi. |
| 13:08.38 | Greenlight | Ahh I was meaning with SIP |
| 13:08.45 | Greenlight | DAHDI can be different |
| 13:09.44 | Greenlight | In fact ISDN30 seems to differ from one call to another |
| 13:09.55 | Greenlight | Sometimes "ringing", sometimes "progress" |
| 13:10.00 | Greenlight | Least with Virgin Media here in UK |
| 13:10.05 | WIMPy | I don;t get anythign from the ITSPs, either. |
| 13:10.24 | Greenlight | You're ITSP's don't generate any ringing ? |
| 13:10.46 | WIMPy | That's probably due to what the called party does. |
| 13:11.19 | WIMPy | NFI who generates the ringin. Never looked in to that. |
| 13:11.36 | WIMPy | I hardly ever use the ITSPs anyway. |
| 13:12.21 | Greenlight | With our AMI application we need to be aware very accurately when the other end has actually started to ring, with DAHDI it was a pain to abstract it all out |
| 13:13.10 | WIMPy | Hmm. Shpuld be pretty obvious. |
| 13:25.14 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 14:16.51 | carrar | moof |
| 14:18.17 | *** part/#asterisk StaRetji (~LittleAll@91.142.129.1) |
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| 14:49.23 | fling | sip guests are calling me but I don't hear anything :| |
| 14:58.40 | igcewieling | Whoo! Whoo! We finally found a customer to take all those horrible Polycom VVX500s we have |
| 15:01.30 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 15:01.46 | carrar | whats wrong with them? |
| 15:03.06 | igcewieling | for one thing the interface is laggy and they like to randomly reboot |
| 15:04.15 | *** join/#asterisk sruffell (~sruffell@nat/digium/x-lyjbrxxwyckxynmz) |
| 15:04.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:04.19 | *** join/#asterisk mmlj4 (1000@ip68-11-55-215.no.no.cox.net) |
| 15:04.36 | igcewieling | I'll take a SoundPoint IP with actual real buttons that actually read your input every time |
| 15:05.01 | mmlj4 | as opposed to which? I missed it |
| 15:05.43 | igcewieling | mmlj4: VVX |
| 15:06.21 | mmlj4 | hey igcewieling... I picked up a gig replacing 650 switches at tulane, factory defective dells, they're paying me to die of boredom, should last 4 months, start next week |
| 15:07.39 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:08.20 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 15:08.39 | mmlj4 | "Premium-quality desktop voice and video solutions" and the buttons don't work? nice |
| 15:09.25 | igcewieling | mmlj4: they don't have many buttons, mostly touch screen |
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| 15:09.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:09.53 | mmlj4 | ah. |
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| 15:19.51 | carrar | nice |
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| 15:56.44 | Katty | hi lads. |
| 15:56.47 | Greenlight | Hmm... say I have in-band DTMF as would be getting passed from PSTN. Is there any way to "strip" that out of band, or prevent the DTMF tones being distinguisable at the other end? |
| 15:56.48 | Katty | and tony. |
| 15:57.12 | jmetro | hi chattykatty |
| 15:57.18 | Katty | waves to jmetro |
| 15:57.22 | Katty | jmetro: how stuff n things |
| 15:57.37 | jmetro | my big ol system i was working on for the past 5 months just went live |
| 15:57.44 | Katty | :>> |
| 15:57.46 | Katty | WOOHOO! |
| 15:57.50 | Katty | fistbump jmetro |
| 15:57.50 | WIMPy | Greenlight: If the receiving channel can do it... |
| 15:57.55 | jmetro | \o/ |
| 15:58.08 | jmetro | only problem is users & our operator panel |
| 15:58.13 | Katty | jmetro: running smoothly, for the most part? |
| 15:58.24 | Katty | which OP you using? |
| 15:58.35 | jmetro | isymphony |
| 15:58.42 | Katty | ah yes. our users like that one too |
| 15:58.50 | Greenlight | WIMPy: Can you explain? Basically I want the call recordings not to have the DTMF tones on them (don't mind a generic tone to replace). It's to do with compliance |
| 15:58.58 | Katty | used to be one of the guys that worked at isymphony hung ou in the channel |
| 15:59.00 | jmetro | its nice except for some really bad parts.. that are so easy to fix but lazy devs |
| 15:59.04 | jmetro | probably myera |
| 15:59.10 | Katty | his name was mike |
| 15:59.12 | jmetro | hes the sole tech support guy |
| 15:59.26 | jmetro | michael yera =p |
| 15:59.33 | WIMPy | Greenlight: Usually if it comes in-band, it stays in-band even if out-of-bad is added. What channel? |
| 15:59.36 | Katty | yup, that's the one! |
| 15:59.40 | Katty | sweet guy. |
| 15:59.51 | Greenlight | Channel ? |
| 15:59.53 | jmetro | he is, but why u no add call duration to call history :< |
| 16:00.04 | WIMPy | channeltype |
| 16:00.10 | Katty | jmetro: suggest it! |
| 16:00.14 | jmetro | i have |
| 16:00.14 | Greenlight | Oh, it'll be SIP or DAHDI |
| 16:00.20 | Katty | jmetro: (= all you can do |
| 16:00.24 | jmetro | isymphony 2 is feature frozen |
| 16:00.29 | Greenlight | I guessed that once in-band it's going to be hard to remove |
| 16:00.49 | Greenlight | I heard other companies replace the DTMF with a single tone |
| 16:01.04 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:01.13 | igcewieling | Asterisk converts inband dtmf on dahdi to rc2833 dtmf all the time |
| 16:01.25 | Greenlight | But doesn't it *stay* inband as well / |
| 16:01.32 | igcewieling | Greenlight: I'm not aware of any way to do what you want to do. |
| 16:01.50 | igcewieling | Greenlight: enable dtmf logging on your asterisk. |
| 16:01.55 | Greenlight | Yea - I didn't think it would be easy |
| 16:02.14 | WIMPy | Oh, even two channels. |
| 16:02.23 | Greenlight | I would need to somehow prevent RTP packets being passed when DTMF starts |
| 16:02.37 | Greenlight | Perhaps a custom function could do it |
| 16:02.57 | WIMPy | That's probably a bit late. |
| 16:03.17 | Greenlight | As long as I can detect before the RTP frame has been passed |
| 16:03.24 | WIMPy | What about postprocessing the recordings? |
| 16:03.48 | Greenlight | The timing would need to be spot on |
| 16:04.07 | jmetro | Katty: and not to mention up until a couple days ago the conference function used meetme |
| 16:04.28 | WIMPy | The thing is that dtmf will always begin before it's detected. So if you mute or replace when it's detected, it will probably be late enough to make it detectable again. |
| 16:04.43 | Greenlight | WIMPy: Damn, that was my fear |
| 16:05.04 | Katty | jmetro: it's still a very easy to use product. that's awesome! |
| 16:05.11 | Greenlight | I guess I could add a few frames worth of latency, but now starting to go down a tricky path |
| 16:05.49 | WIMPy | And you add delay that way. |
| 16:05.55 | Greenlight | Yes, indeed |
| 16:06.10 | Greenlight | But any such solution would have to work like that I'd imagine |
| 16:06.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:06.11 | jmetro | Katty: it is easy to use except when it loses your custom view. |
| 16:06.20 | jmetro | also, whats up with only seeing 1 parking context ? this isnt 1.4 anymore |
| 16:07.05 | WIMPy | Greenlight: Unless it's ok for you to do it after recording. |
| 16:07.44 | Greenlight | WIMPy: Not sure how that would sit with the compliance side; perhaps if it was before it left memory (ramdisK) and before it was accessible by anyone |
| 16:08.04 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 16:09.31 | WIMPy | Can the relevant DTMF come any time or is it just the resonse to a certain promt? |
| 16:09.53 | Greenlight | It would be customers entering their credit card details via their keypads |
| 16:10.23 | WIMPy | So you could pase the recording while that part happens. |
| 16:10.48 | Greenlight | Yea, we currently have that functionality where the agents take the details over the phone and pause the call recoriding |
| 16:10.53 | WIMPy | pause |
| 16:11.31 | Greenlight | But after some length discussions with compliance teams it seems they want even more secure solutions |
| 16:11.48 | Greenlight | I think giving the agents ability to pause recording, opens things up too much |
| 16:13.01 | Greenlight | And apparently there's another company doing this "tone-replace" thing with DTMF |
| 16:13.48 | Greenlight | Will see if I can fish some more details on it. Thanks for the input though, it's appriciated |
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| 16:26.10 | smkelly | file: hi |
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| 16:37.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.231) |
| 16:38.25 | Kobaz | so who wants to help fix a nasty sip problem |
| 16:38.38 | jmetro | sounds like a job for your doctor |
| 16:38.56 | WIMPy | LOL |
| 16:40.16 | Kobaz | insert $$, get fix |
| 16:40.19 | Kobaz | any takers? |
| 16:40.28 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:41.29 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:41.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:41.53 | *** join/#asterisk TimeRider (~steve@0546d828.skybroadband.com) |
| 16:41.58 | [TK]D-Fender | Kobaz: How about you just tell us what the problem is before people start guessing if they can fix it? |
| 16:42.57 | Kobaz | well people need the appropriate skill set first |
| 16:43.07 | Kobaz | anyway... so this is happening very randomly |
| 16:43.19 | Kobaz | phone a is doing a Dial(SIP/b) |
| 16:43.44 | [TK]D-Fender | So acrual debug. Half-assed description is WEAK SAUCE :p |
| 16:43.46 | Kobaz | asterisk sends b the INVITE, b sends back OK.. asterisk waits.... and waits... and then complains it missed a critical packet and hangs up the call |
| 16:43.47 | [TK]D-Fender | actual* |
| 16:43.50 | Kobaz | yeah |
| 16:44.00 | Kobaz | i have lots of debugs, i'll get one |
| 16:44.11 | Kobaz | but that's the gist |
| 16:44.36 | Kobaz | so asterisk is not processing, or is ignoring the OK |
| 16:44.41 | Kobaz | and never sends an ACK to the phone |
| 16:45.12 | igcewieling | Our customers, always trying to make us more profitable " circuit is back up. Someone physically removed the cable connecting the T1 to the router. This was a billable dispatch. " |
| 16:45.25 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:45.49 | mmlj4 | indeed it is |
| 16:45.57 | igcewieling | sounds like typical NAT issue, Kobaz |
| 16:46.07 | Kobaz | you would think it was a nat issue |
| 16:46.12 | Kobaz | but this is all LAN traffic |
| 16:46.35 | Kobaz | and you can clearly see in the sip dump that the server gets the OK |
| 16:46.42 | Kobaz | but asterisk for whatever reason doesnt do anything with it |
| 16:46.43 | igcewieling | unless, of course, asterisk thinks the local LAN is behind NAT based on your sip.conf settings |
| 16:47.00 | jmetro | igcewieling: lol |
| 16:47.05 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:47.21 | igcewieling | Kobaz: remove localnet= externip= externhost= and set directmedia=no and set nat=no |
| 16:47.34 | jmetro | igcewieling: the amount of times ive fixed something by turning it on... i had a user the other day plugged his POE switch into a POE brick and then back into the POE switch. |
| 16:47.37 | Kobaz | anyway, downloading wireshark on this box to extract out the dialog |
| 16:47.38 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.113) |
| 16:47.53 | Kobaz | directmedia is definitely no |
| 16:48.01 | Kobaz | and these phones definitely have nat=no |
| 16:48.03 | Kobaz | what's weird |
| 16:48.10 | igcewieling | what about localnet? |
| 16:48.14 | jmetro | force rport,comedia! |
| 16:48.16 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 16:48.21 | [TK]D-Fender | ...is that you're telling us this and still not showing... |
| 16:48.25 | Kobaz | is if it was a nat type issue, then it wouldn't be so random i would think |
| 16:48.32 | [TK]D-Fender | </storytime> |
| 16:48.34 | [TK]D-Fender | <debug> |
| 16:48.35 | Kobaz | [TK]D-Fender: donwnloading |
| 16:49.05 | igcewieling | Kobaz: if the sky opened up and god himself called me and the call failed, I'd say it was a NAT issue. |
| 16:49.18 | Kobaz | i do have a localnet set |
| 16:49.38 | igcewieling | remove it as well as externip and see if the problem continues |
| 16:49.39 | Kobaz | and it matches the phone subnet... hopefully it wont break anything |
| 16:49.49 | Kobaz | i do have some remote phones |
| 16:49.51 | igcewieling | what was localnet= set to? |
| 16:49.59 | Kobaz | 192.168.50.0/24 |
| 16:50.08 | igcewieling | Kobaz: so what? you are troublehooting right now, they can go down for a while |
| 16:50.14 | Kobaz | haha |
| 16:50.27 | Kobaz | if they go down again i'm screwed |
| 16:50.40 | Kobaz | as in like... attack lawyers |
| 16:51.20 | Kobaz | okay, wireshark installed |
| 16:52.01 | mmlj4 | did you notify them that you were working on it? and they all know there's a ticket out on it, and they were the ones that opened it? drop their calls |
| 16:53.01 | Kobaz | this has been going on for weeks |
| 16:53.07 | Kobaz | lots of people are pissed, etc |
| 16:53.12 | Kobaz | among other problems |
| 16:57.30 | Kobaz | so anyway, gonna take a bit to grab this 50mb sip dump segment |
| 16:57.46 | Kobaz | it doesn't help i'm remote |
| 16:59.24 | mmlj4 | speaking of POE... I found that those little blue switches really, really don't like being plugged into POE :-) |
| 16:59.43 | Kobaz | which blue switches? |
| 16:59.57 | mmlj4 | linksys |
| 17:00.09 | mmlj4 | or linksizzle... |
| 17:00.27 | Kobaz | the blue/black ones? |
| 17:00.32 | Kobaz | i have ports die on those all the time |
| 17:00.55 | mmlj4 | yeah, blue and black |
| 17:01.06 | mmlj4 | POE seems to add more black to them |
| 17:05.31 | *** join/#asterisk LokiScarlet (~loki@crantrap.cranberrytrap.org) |
| 17:06.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.231) |
| 17:06.59 | Kobaz | anyway, one of the vpn links is way slow... so i'll have the dump in a little |
| 17:07.08 | Kobaz | got the file using the better one |
| 17:07.18 | LokiScarlet | Before I get laughed at... Anyone mind an asterisknow user coming in with a question? |
| 17:07.47 | igcewieling | LokiScarlet: unlikely we can help |
| 17:07.53 | blitzrage | LokiScarlet: it's unlikely to get much traction |
| 17:07.54 | Kobaz | we dont know asterisknow |
| 17:08.00 | blitzrage | we use asterisk in a vanilla situation |
| 17:08.03 | Kobaz | so probably you'll get some uneasy silencer |
| 17:08.41 | LokiScarlet | I woulda used it all vanilla buuuuut I'm not the decision maker... And I don't think there's gonna be any activity in #asterisknow... |
| 17:09.20 | igcewieling | Isn't AsteriskNOW a dead product? |
| 17:09.25 | jmetro | i beleive yes. |
| 17:09.41 | jmetro | vanilla + realtime or nothin |
| 17:10.06 | blitzrage | igcewieling: no |
| 17:10.25 | blitzrage | igcewieling: AsteriskNOW 3.0.x was just released and uses Asterisk 11 as the base |
| 17:10.32 | igcewieling | I must be thinking of AsteriskGUI then |
| 17:10.37 | LokiScarlet | ... Okay, I get it, I'm terrible at educating my boss. He's just as spiteful as I am. |
| 17:10.39 | blitzrage | only FreePBX as the interface now though, as the Asterisk-GUI project is no longer receiving development from Digium |
| 17:11.06 | igcewieling | LokiScarlet: ask your question so we can get on with ignoring it |
| 17:11.15 | jmetro | i mean you could always ask, i think the first rule about asking is dont ask if you can ask |
| 17:11.37 | jmetro | ~ask |
| 17:11.37 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:11.37 | jmetro | ? |
| 17:11.47 | leifmadsen | better to ask for forgiveness than for permission |
| 17:12.59 | [TK]D-Fender | leifmadsen: Previously there was a barely-nominal amount of development/support for Asterisk-GUI for some measure of Digium's corporate customers. Is that now completely defunct? |
| 17:13.41 | LokiScarlet | True. I'll get a more detailed error since my boss who likes to take control on the levels he's not educated for, tells me a million different office-jargon ways |
| 17:14.12 | jmetro | synergize the workflow |
| 17:14.22 | Kobaz | [TK]D-Fender: allrightey... www.kobaz.net/misc/dump.pcap |
| 17:14.36 | *** join/#asterisk fish9370 (~Miranda@mbox.fpg.ru) |
| 17:15.30 | fish9370 | hello, who help me with dtmf read? |
| 17:15.57 | WIMPy | fish9370: Ask your question |
| 17:16.15 | [TK]D-Fender | Kobaz: I accept 1 format of debug... you know that.... |
| 17:16.56 | Kobaz | [TK]D-Fender: actually i just found out how to export as text... pastebin: http://pastebin.com/zkQZBn38 |
| 17:17.03 | fish9370 | when I read dtmf, when I pressed 1# asterisk still waiting button |
| 17:17.03 | fish9370 | why? |
| 17:17.05 | LokiScarlet | Okay so that was embarrassment for nothing. IT directors, I hate them so much. His anecdote matches none of the data. Sorry bout all that |
| 17:17.55 | LokiScarlet | (This is how I now find out that if I question his anecdotes and do my job, I start fights :P ) |
| 17:18.02 | fish9370 | but if I pressed 1#<some digit> it's break |
| 17:18.33 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:18.36 | Kobaz | [TK]D-Fender: server is 192.168.51.1, sip device is 192.168.51.20 |
| 17:18.52 | [TK]D-Fender | Kobaz: * SIP debug + verbose 10 |
| 17:18.54 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 17:19.07 | [TK]D-Fender | Kobaz: I don't do substitutes. |
| 17:19.08 | jimi_ | Is/wasn't there a command to show moduletypes? i tried core show moduletypes but that wasnt it |
| 17:19.33 | [TK]D-Fender | fish9370when I read dtmf, when I pressed 1# asterisk still waiting button <- read where? |
| 17:19.50 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:19.53 | [TK]D-Fender | fish9370: You are completely vague as to the circumstances... |
| 17:20.12 | fish9370 | in agi get_data (function ast_readstring_full) |
| 17:21.01 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:24.39 | Katty | ASTERISK. YEAH. |
| 17:24.52 | [TK]D-Fender | fish9370: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AGICommand_get+data <- I don't see any mention of "#" being a terminator at all |
| 17:24.54 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:24.59 | Katty | hi fender bender |
| 17:25.35 | fish9370 | [TK]D-Fender: I look at code, in function ast_readstring_full(struct ast_channel *c, char *s, int len, int timeout, int ftimeout, char *enders, int audiofd, int ctrlfd) |
| 17:25.43 | fish9370 | here is enders |
| 17:26.02 | danfromuk | Hi. Where can I find an explanation of the different options of the 'events' AMI command? |
| 17:26.05 | fish9370 | enders it's combination for end reading |
| 17:26.27 | Katty | Qwell: o/ |
| 17:27.03 | Qwell | Katty: ohai |
| 17:27.06 | Katty | how'rechu dear |
| 17:27.11 | Qwell | I be! |
| 17:27.17 | Katty | and the ladyfriend, of course. |
| 17:27.47 | Katty | eggscelent. glad you are being. |
| 17:27.59 | Qwell | Katty: We're driving to OK tonight. |
| 17:27.59 | jmetro | Doke, OK? |
| 17:28.01 | Katty | oooh shenanigans! my favorite! |
| 17:28.05 | fish9370 | and when it's called, it's called as ast_readstring_full(c, s, maxlen, to, fto, "#", audiofd, ctrlfd); |
| 17:28.13 | Katty | Qwell: what's on the OK agenda |
| 17:28.19 | Katty | is always looking for new places to go |
| 17:28.21 | fish9370 | where # is't enders |
| 17:28.49 | fish9370 | [TK]D-Fender: ? |
| 17:29.49 | WIMPy | danfromuk: The wiki |
| 17:29.49 | danfromuk | WIMPy: its not there. Just gives a list but no explaination. |
| 17:29.56 | WIMPy | So what exactely are you looking for? |
| 17:30.25 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 17:30.28 | danfromuk | Just want to understand the different options to ensure i'm excluding what i dont need. |
| 17:30.41 | fish9370 | danfromuk: look at code |
| 17:31.31 | LokiScarlet | Alright. Let's just say I'm using Asterisk and FreePBX on a CentOS6 box, using the asterisk and digium repos for CentOS. I can't seem to get a hold of the chan_console.so module and I dare not try to recompile asterisk manually with an IT director who learned how to use computers from Dell |
| 17:40.08 | [TK]D-Fender | fish9370: I don't know about your code but I'd go look at the AGI debug to verify what's actually getting called, and then prove that DTMF even works outside that |
| 17:40.20 | Katty | fender. |
| 17:40.50 | [TK]D-Fender | Katty: Mew. |
| 17:41.17 | Katty | how'rechu dear |
| 17:41.40 | fish9370 | [TK]D-Fender: what for you need my code, when handler in asterisk core? |
| 17:41.55 | [TK]D-Fender | Katty: Busy bee... new bad in final weeks before gigging starts. |
| 17:42.19 | [TK]D-Fender | Katty: 1st show in 2 weeks, then back at the end of august and then mid-october |
| 17:42.21 | fish9370 | [TK]D-Fender: in chanel.c |
| 17:42.47 | Katty | [TK]D-Fender: yay |
| 17:42.53 | [TK]D-Fender | fish9370: Not talking about * source". Go prove what AGI call is really being made and debug the calling channel itself |
| 17:44.00 | fish9370 | [TK]D-Fender: ? |
| 17:44.23 | Qwell | Katty: We're driving to OK tonight. |
| 17:44.28 | Qwell | erm, this isn't my console |
| 17:45.06 | Katty | Qwell: visiting any cool stuff out there? i'm always looking for new things to do |
| 17:45.26 | Qwell | Katty: not in Tulsa, no. We're going to Dallas afterwards though. |
| 17:45.48 | [TK]D-Fender | [13:20]fish9370in agi get_data (function ast_readstring_full) |
| 17:45.58 | Katty | dallas was fun. |
| 17:46.03 | [TK]D-Fender | fish9370: You said you were using an AGI. So show us the call debug with AGI debug enabled |
| 17:46.07 | Katty | nice little aquarium in the middle of the city. |
| 17:46.33 | Qwell | Katty: Yes, Dallas World Aquarium. We are going. |
| 17:46.35 | fish9370 | how? it's so long |
| 17:46.43 | Katty | Qwell: you'll love it! |
| 17:46.51 | Katty | Qwell: bit chilly. take a hoodie |
| 17:46.59 | Qwell | Katty: well, I drive to Chattanooga every other weekend or so. |
| 17:47.01 | fish9370 | [TK]D-Fender: what you want to see |
| 17:47.04 | fish9370 | ? |
| 17:47.08 | Qwell | I'm pretty much set in the aquarium dept. |
| 17:47.15 | Katty | wait. |
| 17:47.18 | *** join/#asterisk Olobola (Olobola@c-71-198-191-166.hsd1.ca.comcast.net) |
| 17:47.19 | [TK]D-Fender | fish9370: What part of "call debug" is unclear? |
| 17:47.25 | Katty | aqarium? chattanooga? |
| 17:47.30 | Katty | details. |
| 17:47.40 | [TK]D-Fender | fish9370: Asterisk CLI, verbose 10, AGI debug enabled, and any channel debug pertinent to the device placing the call. |
| 17:47.53 | LokiScarlet | An aquarium in Dallas? Chilly? Tell that to the coldest day of winter, in shorts. |
| 17:48.09 | fish9370 | [TK]D-Fender: ok, 5 sec |
| 17:48.10 | Qwell | Katty: Tennessee Aquarium is at the top of the list, of best aquariums. |
| 17:48.49 | Katty | that's not the gatlinburg one, is it? |
| 17:48.59 | Qwell | Katty: no, downtown Chattanooga |
| 17:49.08 | Katty | researches |
| 17:49.30 | LokiScarlet | Qwell: Just saying, Tennessee has some tourist attractions but I feel sorry for you if you're in or move to this state. |
| 17:49.44 | Qwell | LokiScarlet: No, no, much worse. Alabama. |
| 17:49.51 | fish9370 | [Jun 7 21:30:39] <SIP/5992-00000022>AGI Rx << GET DATA /var/lib/asterisk/sounds/ru/NovieGrani 5000 3 |
| 17:49.51 | fish9370 | [Jun 7 21:30:39] -- <SIP/5992-00000022> Playing '/var/lib/asterisk/sounds/ru/NovieGrani.slin' (language 'ru') |
| 17:49.51 | fish9370 | [Jun 7 21:30:47] <SIP/5992-00000022>AGI Tx >> 200 result=1 (timeout) |
| 17:49.51 | fish9370 | [Jun 7 21:30:47] <SIP/5992-00000022>AGI Rx << VERBOSE "SIP/5992-00000022 -- . buttons pressed 1" 1 |
| 17:49.51 | fish9370 | [Jun 7 21:30:47] in.php: SIP/5992-00000022 -- . buttons pressed 1 |
| 17:49.51 | fish9370 | [Jun 7 21:30:47] <SIP/5992-00000022>AGI Tx >> 200 result=1 |
| 17:50.02 | Katty | Qwell: holy bluespotted sunfish batman! |
| 17:50.08 | fish9370 | [TK]D-Fender: it's ok? |
| 17:50.09 | WIMPy | ~pb |
| 17:50.09 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:50.10 | Katty | adds chattanooga on her to do list |
| 17:50.20 | Katty | Qwell: any other places of note out there? |
| 17:50.31 | Katty | Qwell: or tasty resturants. breweriers, and the like |
| 17:50.34 | LokiScarlet | Ow. I feel your pain. I'm looking at houses in Pulaski and already feeling like that's waaay too far south. But work is work and land is cheap right now |
| 17:51.16 | fish9370 | [TK]D-Fender: I pressed 1# but how you can see it's timed out |
| 17:52.17 | WIMPy | fish9370: That looks like you try to read before the call has even been answered. |
| 17:52.36 | fish9370 | WIMPy: no |
| 17:53.23 | [TK]D-Fender | fish9370: now go prove that SIP/5992 works with anything else using DTMF. |
| 17:53.30 | fish9370 | if ($answer) |
| 17:53.30 | fish9370 | $agi->answer(); |
| 17:53.30 | fish9370 | $buttons = $agi->get_data($filename, 5000, $count); |
| 17:53.37 | fish9370 | there is logic |
| 17:53.46 | [TK]D-Fender | fish9370: and pastebin the COMPLETE call <------- |
| 17:53.48 | [TK]D-Fender | ^ |
| 17:53.50 | fish9370 | $answer = true |
| 17:55.44 | fish9370 | [TK]D-Fender: ? |
| 17:55.59 | [TK]D-Fender | fish9370: what is unclear now? |
| 17:56.27 | fish9370 | [TK]D-Fender: I don't understand, how prove |
| 17:56.29 | [TK]D-Fender | fish9370: PASTEBIN ..... the COMPLETE call. |
| 17:56.56 | [TK]D-Fender | fish9370: and go use something OTHER than AGI to prove that * is even getting DTMF from the other end |
| 17:56.58 | fish9370 | what is PASTEBIN? |
| 17:57.09 | WIMPy | ~pb |
| 17:57.09 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:57.11 | [TK]D-Fender | fish9370: Call Read() from the dialplan, Voicemail(), or whatever... |
| 17:57.23 | [tpn]leifmadsen | ProTip: never use While() if you plan to nest subroutines that could call themselves |
| 17:57.35 | fish9370 | oh, ok ) |
| 17:57.51 | fish9370 | sorry |
| 17:58.05 | [TK]D-Fender | [tpn]leifmadsen: I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I... |
| 17:58.07 | [TK]D-Fender | ...do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't always recurse, but when I do I don't... |
| 17:58.08 | [TK]D-Fender | ...always recurse, but when I do I don't always recurse, but when I do |
| 17:58.16 | [TK]D-Fender | :) |
| 17:59.32 | Katty | asplodes. |
| 18:00.10 | fish9370 | ok, I find bug, can I say about it? |
| 18:00.17 | fish9370 | other bug |
| 18:00.26 | [TK]D-Fender | fish9370: Don't ask if you can ask. |
| 18:00.35 | fish9370 | ok, |
| 18:00.39 | [TK]D-Fender | fish9370: fish9370 Just get it out... |
| 18:01.06 | fish9370 | if in dialplan we have template like _X! |
| 18:02.04 | fish9370 | and we want transfer call throw this template we missing call |
| 18:02.14 | fish9370 | it's bug |
| 18:02.17 | [TK]D-Fender | that is a pattern, not a "template", and no.... |
| 18:02.27 | fish9370 | yes, yes sorry |
| 18:02.33 | [TK]D-Fender | it's not a bug. You have not proven your complete code and shown us the actual failed attempt. |
| 18:02.43 | WIMPy | What do you think is a bug? |
| 18:02.49 | [TK]D-Fender | Because it doesn't work! |
| 18:02.58 | [TK]D-Fender | It couldn't possibly be user error! |
| 18:03.02 | WIMPy | You're throwing pretty incomplete stuff at us. |
| 18:03.13 | fish9370 | becouse _X! it's 1 or any digits, yes? |
| 18:03.30 | fish9370 | right? |
| 18:03.34 | WIMPy | At least one digit. |
| 18:03.35 | [TK]D-Fender | fish9370: We have NO IDEA where you are talking about that "_X!" even occuring |
| 18:03.41 | [TK]D-Fender | fish9370: SHOW US! |
| 18:03.43 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 18:03.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 18:04.01 | WIMPy | Or what you're trying to match or when/from where. |
| 18:05.27 | *** join/#asterisk kresp0 (~kresp0@35.Red-83-39-25.dynamicIP.rima-tde.net) |
| 18:07.33 | fish9370 | i can't explain |
| 18:07.51 | fish9370 | english not so good |
| 18:08.48 | fish9370 | who speek russian? |
| 18:10.30 | WIMPy | It's not the language, it's the fact that your descriptions are extremely incomplete so we don;t really know what your question is. |
| 18:10.58 | fish9370 | WIMPy: i understand |
| 18:11.06 | fish9370 | WIMPy: you right |
| 18:12.47 | fish9370 | i try provide proof |
| 18:13.13 | WIMPy | Better tell us what exactely you're trying to do. |
| 18:14.02 | fish9370 | i receive call, answer, and put it in context with pattern _X! |
| 18:14.13 | fish9370 | no, wait |
| 18:14.26 | WIMPy | You don't put a call in to a context after answering. |
| 18:14.27 | [TK]D-Fender | fish9370: Go do NORMAL dialplan and use the READ() dialplan application and prove that SIP/5992 |
| 18:14.45 | fish9370 | i receive call, answer, and transfer call int context with pattern _X! |
| 18:14.46 | WIMPy | Unless you use some redirect action. |
| 18:14.55 | [TK]D-Fender | fish9370i receive call, answer, and put it in context with pattern _X! <------ show us your code and the call. |
| 18:15.05 | [TK]D-Fender | fish9370: Stop saying "it doesn't work", and show us. |
| 18:15.25 | WIMPy | Using a feature (inband dtmf) transfer? |
| 18:15.39 | WIMPy | And what happens and what do you expect to happen? |
| 18:15.40 | fish9370 | WIMPy: yes, it's after transfer |
| 18:16.38 | fish9370 | WIMPy: i expect call go throw this pattern.. but i have hangup |
| 18:17.13 | WIMPy | Are you sure you end up in the right context? |
| 18:17.27 | WIMPy | Turn up verbose and show us what happens. |
| 18:17.36 | [TK]D-Fender | well ... I've certainly wasted enough time on this... |
| 18:17.39 | fish9370 | WIMPy: but, if i replace this pattern with _X. it's work |
| 18:17.51 | [TK]D-Fender | moves on to more productive matters |
| 18:18.23 | fish9370 | WIMPy: it's normal? |
| 18:18.54 | WIMPy | If that's the only pattern you have in that context, it will match as soon as you dial the first digit. That's the idea. |
| 18:19.45 | fish9370 | WIMPy: yes, you right |
| 18:20.29 | fish9370 | WIMPy: if i make outbound call throw this pattern like 100 it match |
| 18:20.35 | WIMPy | If that's not what you want use patterns that do what you want. |
| 18:21.16 | fish9370 | WIMPy: yes, i can, but it's wrong |
| 18:21.46 | WIMPy | What is wrong? |
| 18:21.47 | fish9370 | WIMPy: then i can't handle one digit |
| 18:22.15 | fish9370 | becouse _X. it's two or more |
| 18:22.27 | WIMPy | Use two patterns. _X and _X. . |
| 18:22.52 | fish9370 | then i need two contexts, but why? |
| 18:23.04 | WIMPy | No, two extensions in one context. |
| 18:23.12 | fish9370 | _X! one or more |
| 18:23.22 | WIMPy | That's not the only difference. |
| 18:23.47 | fish9370 | two extensions, it's to much |
| 18:23.54 | fish9370 | *too |
| 18:24.15 | WIMPy | ! also means to match as soon as possible. Do no use ! if you want to be able to continue dialling, unless you pass it on to a channel that can handle overlap dialling. |
| 18:24.19 | fish9370 | it's wrong |
| 18:25.14 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
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| 18:25.29 | fish9370 | thanks, WIMPy, i go home.. |
| 18:25.32 | fish9370 | bye |
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| 20:15.25 | jmetro | these autodestruct messages are killing me |
| 20:15.41 | jmetro | i have no Dials with a g |
| 20:15.46 | jmetro | just getting spammed |
| 20:20.11 | pabelanger | jmetro: failed to autodestruct or something like that? |
| 20:20.16 | jmetro | yeah |
| 20:20.19 | jmetro | requesting BYE etc etc |
| 20:20.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hmcjfhiwkypijryp) |
| 20:20.43 | [TK]D-Fender | And how are they "killing you"? |
| 20:20.45 | pabelanger | jmetro: make sure you are not blocking the asterisk thread.... eg: an AGI script or system call. Had that issue in the past |
| 20:21.03 | jmetro | [TK]D-Fender: after a bunch of them stack up i actually get dialplan malfunctions |
| 20:21.08 | pabelanger | System(sleep 120) is bad |
| 20:21.09 | jmetro | perfectly normal dialplan just fails |
| 20:21.22 | [TK]D-Fender | jmetro: That is a special kind of awesome.... |
| 20:22.17 | jmetro | Lately its only been for one user - dialing her find/follow will result in dead air on the cellphone for her, or dead-air on the phone when it reaches her vm |
| 20:24.05 | jmetro | only for these two users apparently now |
| 20:25.19 | LokiScarlet | Hi. Been googling, I know this has been opened and closed for someone else, but I can't seem to get working what they got working. I seem to get permission denied for the sound device when loading chan_alsa. Asterisk is in the audio group. |
| 20:25.45 | LokiScarlet | (Trying to be as quick and detailed as possible) |
| 20:27.30 | LokiScarlet | TLDR I'm a stupid noob and I'm getting permission denied loading the alsa channel |
| 20:27.59 | jmetro | [TK]D-Fender: the autodestructs seem to not happen for a long time, then randomly they start, and its just like once one happens they just STACK UP LIKE MAD |
| 20:28.49 | jmetro | 3 or 4 messages every 2 seconds |
| 20:29.43 | *** join/#asterisk bn-7bc (~bjarne-im@2001:16d8:ee6c:1:12dd:b1ff:febc:87ff) |
| 20:30.46 | [TK]D-Fender | :/ |
| 20:30.50 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:32.45 | jmetro | quick question |
| 20:32.59 | jmetro | if i have an announcement on dial eg Dial(SIP/100,a(mymacro)) |
| 20:33.07 | jmetro | and at the end of macro-mymacro i have a hangup |
| 20:33.16 | jmetro | will that fudge the call and hang up stuff that shouldnt be hung up? |
| 20:33.32 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip503c898d.speed.planet.nl) |
| 20:37.00 | WIMPy | There's something wrong with that question. |
| 20:37.23 | WIMPy | a doesn't take parameters and A takes a file. Where is the macro coming from? |
| 20:37.34 | jmetro | ah... |
| 20:37.36 | jmetro | i mean M |
| 20:37.46 | jmetro | m is for macro, thats good enough for me |
| 20:38.38 | WIMPy | m doesn't exist any more. |
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| 20:39.53 | jmetro | M, capital |
| 20:40.40 | WIMPy | Not found |
| 20:42.14 | jmetro | <.< |
| 20:42.48 | jmetro | same => n,Dial(${ARG1},20,TKktrM(announceIncoming,${CutExten})) |
| 20:42.52 | jmetro | like that. |
| 20:44.08 | WIMPy | Looks like that won't work after the next update. |
| 20:44.20 | WIMPy | Better use gosubs. |
| 20:44.30 | jmetro | :< |
| 20:44.57 | jmetro | so i have to make a separate sound file for all my announces |
| 20:45.20 | jmetro | thats...srsly? |
| 20:45.22 | WIMPy | No, use the gosub options instead of the old macro versions. |
| 20:45.52 | jmetro | its already a sub, the point is that i want code to be run when the callee picks up that determines their announcement |
| 20:46.31 | jmetro | getting rid of M is like deciding Dial() cant have a time option |
| 20:46.45 | jmetro | since you could just time it with the system clock or some crap |
| 20:47.27 | WIMPy | Or removing the ENUM switch? |
| 20:48.34 | WIMPy | Doh |
| 20:48.54 | jmetro | so considering i already have like 5 macros deep |
| 20:49.02 | jmetro | i guess i gotta go 6 just to pop off an announcement with my dial |
| 20:49.19 | WIMPy | M *is* still there. But macros have been depercated in favour of gosub anyway. |
| 20:49.49 | jmetro | im totally okay with getting rid of the "macro-" part but theres gotta be a way to tell dial to use it |
| 20:50.19 | WIMPy | U |
| 20:50.35 | jmetro | i just treat macros as subroutines anway |
| 20:52.48 | [tpn]leifmadsen | jmetro: with M() it's ^ as a separator |
| 20:53.17 | WIMPy | Same with U. |
| 20:53.32 | [tpn]leifmadsen | +1 |
| 20:53.35 | jmetro | i dont have any ^'s in my code. |
| 20:53.40 | [tpn]leifmadsen | there's your problem :) |
| 20:53.55 | jmetro | everything works? <.< |
| 20:54.04 | WIMPy | Nooooooo |
| 20:54.22 | jmetro | if i dint have ^'s wouldnt it be broke |
| 20:54.26 | WIMPy | But I guess it should do what you want there :-) |
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| 20:58.28 | jmetro | same => n(ShortCell),Dial(Local/${EXTEN}@mycompany/n,20,tkM(announce-Incoming,${EXTEN})) |
| 20:58.34 | jmetro | my dialer for cells with short ringers |
| 20:58.51 | jmetro | shouldnt that be announce-incoming^${EXTEN}? it works as it is so.. |
| 20:58.55 | WIMPy | And still the wrong syntax. |
| 20:58.58 | WIMPy | indeed |
| 20:59.05 | jmetro | the console never complains when i reload it |
| 20:59.07 | [tpn]leifmadsen | +1 |
| 20:59.17 | [tpn]leifmadsen | you're doing it wrong |
| 20:59.45 | WIMPy | There are only so many things Asterisk complains about. |
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| 21:00.13 | [tpn]leifmadsen | M(macro[^arg[^...]]): |
| 21:00.18 | jmetro | ^ is such an odd delimeter |
| 21:00.19 | [tpn]leifmadsen | from 'core show application Dial" |
| 21:00.35 | [tpn]leifmadsen | jmetro: it's different because , is for the Dial() separator |
| 21:00.44 | [tpn]leifmadsen | so you're nesting, thus you need a different separator |
| 21:00.51 | [tpn]leifmadsen | it's weird because it's unlikely to conflict with another separator |
| 21:00.59 | [tpn]leifmadsen | it's weird on purpose |
| 21:01.03 | jmetro | hm |
| 21:01.09 | [tpn]leifmadsen | so again, you're doing it wrong ;) |
| 21:01.12 | WIMPy | You don;t need it. If the parser was just a tiny bit cleverer it would find the (). |
| 21:01.23 | jmetro | if it was a problem wouldnt things not work? |
| 21:01.28 | [tpn]leifmadsen | probably |
| 21:01.32 | [tpn]leifmadsen | it's still wrong ;) |
| 21:01.38 | jmetro | cause, everything works, and the var passes properly |
| 21:01.50 | [tpn]leifmadsen | possibly due to some lucky parsing |
| 21:01.51 | [tpn]leifmadsen | I wouldn't rely on it |
| 21:03.06 | jmetro | lucky parsing ಠ_ಠ |
| 21:03.07 | mmlj4 | don't listen to [tpn]leifmadsen, all he did was write books about this stuff |
| 21:03.12 | [tpn]leifmadsen | I know nothing! |
| 21:03.16 | [tpn]leifmadsen | I make shit up ALL THE TIME |
| 21:04.41 | jmetro | Good luck jmetro gets lucky parsing ( ͡° ͜ʖ ͡°) |
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| 22:23.52 | EyePulp | *howdy* =) |
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| 22:28.44 | Surye | Hello, we are using a SIP trunk providor, and we have another site (connected via a VPN tunnel), with multiple phones. We're using ASA5505s in both places, and the PBX communicates with phones on the other end of the VPN |
| 22:29.20 | Surye | Is there any way to get the media RTP traffic for the phone on the other site to use their internet, and not use the tunnel after the session is negotiated? |
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