| 00:01.58 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 00:34.53 | apb1963 | so I dial my google DID... my IVR answers. I dial ext 101 which is my phonerlite softphone which is registered and waiting for a call. Asterisk gives me a busy signal. The logfile has a line which says DEVICES=104 as well as THISDIAL=SIP/104... I would think this would be relevant, but I don't know where to go from here. To my still untrained eye, extensions.conf seems ok. |
| 00:35.23 | apb1963 | Any ideas on what I messed up? |
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| 00:55.15 | [TK]D-Fender | apb1963: last log you showed clearly showed that 104 was dialed, not 101 |
| 00:55.50 | apb1963 | yeah... yet I dialed 101... I watched my fingers |
| 00:57.20 | [TK]D-Fender | Go do it again. |
| 00:57.37 | [TK]D-Fender | Beacues there is little reason to trust a human over a log telling me otherwise |
| 00:58.19 | apb1963 | I just did |
| 00:58.28 | apb1963 | log coming up |
| 00:58.36 | [TK]D-Fender | Correct. Little reason to trust you. |
| 00:58.38 | [TK]D-Fender | Do it again |
| 00:58.42 | [TK]D-Fender | Test it from another source |
| 00:58.51 | apb1963 | I don't have another source |
| 00:58.52 | [TK]D-Fender | try to be complete about your testing |
| 00:58.56 | [TK]D-Fender | use a PHONE |
| 00:59.01 | apb1963 | I am using a phone |
| 00:59.04 | [TK]D-Fender | ... |
| 00:59.07 | [TK]D-Fender | an EXTENSION |
| 00:59.09 | apb1963 | this entire time |
| 00:59.11 | [TK]D-Fender | direct to the IVR |
| 00:59.18 | apb1963 | softphone? |
| 00:59.22 | [TK]D-Fender | ANYTHING |
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| 01:06.53 | apb1963 | ok, I can't dial myself using google to dial out - which is my only option for dialing out with my softphone. I use google for both incoming and outgoing when using Asterisk. To make a normal call, I use my normal phone, which is tied into the PSTN. Those are my only options. |
| 01:07.38 | [TK]D-Fender | .... |
| 01:07.49 | [TK]D-Fender | use the softphone. Got to the IVR |
| 01:07.49 | apb1963 | So... when I use my normal phone to make a normal phone into asterisk, it does what I showed you in the log. |
| 01:07.50 | [TK]D-Fender | DIRECTLY |
| 01:09.02 | [TK]D-Fender | While you're at it, do MORE DTMF testing with your GV inbound. |
| 01:09.04 | apb1963 | OK, I can setup my softphone to register me as ext 1001, and then connect to 101... is that what yo're saying to do? |
| 01:09.12 | [TK]D-Fender | And remember that would you use to call GV could be a factor as well |
| 01:09.35 | [TK]D-Fender | Dear God... |
| 01:09.36 | apb1963 | oh the IVR.. right... nvm the abovfe |
| 01:11.19 | apb1963 | ok... I don't know how to go to the ivr through the softphone without first dialing the gv number. |
| 01:11.40 | apb1963 | I can dial an extension... but... what extension is the ivr? |
| 01:11.46 | apb1963 | do I just press... 1 ? |
| 01:12.02 | apb1963 | yeah, I don't follow ya |
| 01:13.22 | apb1963 | now if I register as ext 1001 and dial 101, then I get a proper vm as expected. |
| 01:13.45 | apb1963 | but I don't know how to dial IVR |
| 01:15.02 | [TK]D-Fender | Custom Destination <- |
| 01:15.49 | apb1963 | so I would have to set that up then |
| 01:15.55 | [TK]D-Fender | apb1963: Go prove your DTMF is proper in from GV, and with every tool used in your tests |
| 01:16.16 | [TK]D-Fender | apb1963: Your log shows something else was dialed.. either you screwed up or DTMF is picked up wrong somewhere |
| 01:16.24 | [TK]D-Fender | And you can bet I bet on human failure |
| 01:16.31 | [TK]D-Fender | Show a NEW CALL |
| 01:16.41 | [TK]D-Fender | And try to be thorough about your testing |
| 01:16.42 | apb1963 | the reason I became aware of this problem is someone emailed me to let me know they couldn't get through. |
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| 01:20.51 | sheebacatfood | speaking of google, no tutorial on the internet correctly shows you how to enable your voicemail using a gv trunk. all tutorials are terrible |
| 01:21.26 | sheebacatfood | http://forum.xda-developers.com/showthread.php?t=2117336 (#5 - Voicemail) Wrong. doesn't work. |
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| 01:23.54 | sheebacatfood | http://nerdvittles.com/?tag=vm (Enabling Google Voicemail) Wrong. Doesn't work at all. |
| 01:24.10 | sheebacatfood | is there a single person on the internet who actually knows what they're doing with Asterisk? I don't think there is. |
| 01:26.11 | [TK]D-Fender | You must have the best stats to back that then.... |
| 01:28.58 | [TK]D-Fender | As for "it" not working .... ok/fine/sure |
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| 01:35.46 | apb1963 | ok, I think I may know what's happening.. IVR Option 1 is set to route to ext 104. Before I can finish dialing 101, it picks off the first 1 and routes it to 104. |
| 01:37.05 | apb1963 | So... I guess I need to somehow slow down the "recognition" of the options.... allow more time for the whole extension to be dialed. |
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| 01:40.51 | apb1963 | See for yourself... http://ix.io/5sg |
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| 01:46.28 | [TK]D-Fender | apb1963: It acts on that first DTMF instantly |
| 01:46.42 | [TK]D-Fender | apb1963: Which pretty much means you didn't ENABLE direct extension dialing |
| 01:46.44 | apb1963 | yes... that's what I said |
| 01:46.58 | apb1963 | hhmm... that I will doublecheck |
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| 01:52.27 | apb1963 | Well, I'm going to say it's because I didn't enable direct extension dialing... which is really weird, because this used to work... So clearly it's a gremlin issue. |
| 01:52.38 | apb1963 | damn gremlins |
| 01:53.32 | [TK]D-Fender | Next time check your work :p |
| 01:53.37 | apb1963 | Thank you [TK]D-Fender =:) |
| 01:53.46 | apb1963 | If I knew what to look for... |
| 01:54.00 | apb1963 | I would |
| 01:54.09 | [TK]D-Fender | "I can't dial my extension ....maybe I should check the thing that's supposed to LET me ..." |
| 01:54.30 | apb1963 | meh. One has to know there's something that's supposed to let me |
| 01:54.34 | [TK]D-Fender | Enter ... MAGIC CHECK-BOX!!!! |
| 01:55.19 | apb1963 | I mean if there were like 5 checkboxes total... ok fine.... but there's options upon options that have baby rabbit options. It's madness. madness I tell you |
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| 01:57.23 | apb1963 | and why isn't there a madness emoticon? |
| 01:58.48 | [TK]D-Fender | apb1963: it's on back-order, right behind the "Hides face in shame at not being able to manage an idiot-accomodating GUI" emoticon :) |
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| 01:59.46 | [TK]D-Fender | apb1963: Well at least one more problem solved.... |
| 02:02.18 | apb1963 | looks at the channel name.... |
| 02:02.24 | apb1963 | GUI? What GUI? :P |
| 02:02.56 | [TK]D-Fender | smirks |
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| 02:24.18 | Mango45 | Anyone know what this USB->RJ11 adapter is used for? http://www.vpi.us/usb-rj45.html |
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| 02:38.39 | sheebacatfood | those of you with voicemail working on incoming calls in FreePBX can you give us the code in extensions_additional.conf? nothing I've tried is working. All information on every forum is bad and/or outdated. |
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| 03:07.09 | ChannelZ | welcome to FreePBX! |
| 03:08.23 | ChannelZ | Where the Free means 'Free From Documentation' |
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| 03:08.34 | tm1000 | ChannelZ: can you stop |
| 03:08.45 | tm1000 | ChannelZ: we have PLENTY of documentation |
| 03:08.52 | tm1000 | ChannelZ: http://wiki.freepbx.org |
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| 03:09.56 | ChannelZ | I was being both ironic AND sarcastic |
| 03:10.33 | tm1000 | ChannelZ: /me is sad panda. but ok :-) |
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| 03:57.34 | phix | heh |
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| 05:26.50 | supercell | not asterisk related, but I bouhgt a did. When I try to call it from cell phone I get a msg from verizon switch unavailable or something |
| 05:30.25 | supercell | --any ideas? |
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| 05:44.59 | [TK]D-Fender | supercell: Well .. it isn't asterisk related... and you've told us pretty much nothing about what you;re doing and shown us the same. Nothing for us to help you with |
| 05:46.23 | [TK]D-Fender | [22:38]sheebacatfoodthose of you with voicemail working on incoming calls in FreePBX can you give us the code in extensions_additional.conf? nothing I've tried is working. All information on every forum is bad and/or outdated. <- there is no coding to do. It's all done in the GUI |
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| 05:59.11 | supercell | I guess I was just looking for generic information about the DID purchase process, and why one carrier might not connect to another etc |
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| 06:00.42 | [TK]D-Fender | supercell: You bought a DID... it's supposed to GO somewhere. So far you haven't said that you told them what to do with it. So they have nothing to do and say "WTF now?" |
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| 06:15.57 | [TK]D-Fender | apb1963: Bed time, I'm off... |
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| 07:13.54 | caterwaul | does anyone compile their own CSipSimple by chance? |
| 07:20.16 | kaldemar | do they have their own support channel(s)? |
| 07:23.16 | emk_ | ls |
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| 08:04.59 | bulkorok | hi |
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| 11:46.53 | janelleb | hey all, I'm having issues handling SMS... asterisk is dealing with received sms's as if they were calls. I'm using almost the stock/vanilla extensions.conf (http://pastebin.com/YuDcwt5N), on asterisk 1.8 with chan_dongle. How can I have a simple "rule" like "when asterisk receives an sms, execute this_dialplan_application"??? |
| 11:55.46 | Tuju | can asterisk process sms messages? |
| 11:58.33 | janelleb | Tuju: I think so... it can atleast do_someting in response to an sms... i.e. pass the SMS on to curl or something. This is what I've read online but I'm working on trying to do it right now. |
| 11:59.12 | Tuju | well that's interesting. |
| 11:59.45 | Tuju | i just learned what the MWI is and that sounded also something worth of trying, thou i don't like voice messages. |
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| 13:42.15 | Greenlight | Is it normal that when I hangup a call via the AMI, the Hangup event that follows shows the hangup cause as "Unknown" ? |
| 13:43.02 | SuperNull | Hey guys, what is the expected delay between a message file being written to the voicemail directory and the message waiting indicator being generated for the related user ? |
| 13:43.16 | igcewieling | SuperNull: a few seconds |
| 13:43.25 | SuperNull | hurm. |
| 13:44.16 | SuperNull | oddly i can get calls through nat, but damn MWI doesn't clear. a keep alive is even being sent to keep the translation active. |
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| 13:49.49 | Katty | infobot: crittercam |
| 13:49.49 | infobot | i guess crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
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| 13:50.38 | atan | Katty, your street signs looks like what I have around here... where do you live? |
| 13:51.14 | atan | Hmm. Text on page gives it away. I wonder if you're local to me. Hmmmmmm cute birds! |
| 13:51.15 | Katty | missouri (= |
| 13:51.33 | Katty | ty. |
| 13:51.42 | Katty | hopefully the squirrels will be about soon. |
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| 14:11.55 | RadJackson | Hello , i make some automatic phonecalls , by placing ".call" files into /var/spool/asterisk/outgoing folder, every minute asterisk launch one file, only 30% of the calls works, the other 70% displays an error message saying pbx_spool.c: Call failed to go through, reason (3) Remote end Ringing / devicestate.c: No provider found, checking channel drivers for SIP- XXX |
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| 14:14.11 | kaldemar | RadJackson: did you look at the sip debug for such a call? |
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| 14:16.09 | RadJackson | Sorry , i am new to Asterisk , how can i see the sip debug? |
| 14:16.46 | RadJackson | through the CLI it says failed to go through, reason (3) Remote end Ringing, and on the full log file it says No provider found, checking channel drivers for SIP- XXX |
| 14:17.24 | WIMPy | SuperNull: None |
| 14:18.40 | kaldemar | RadJackson: you enable it in asterisk's CLI with "sip debug" |
| 14:18.55 | kaldemar | sorry, "sip set debug on" |
| 14:19.05 | kaldemar | "sip set debug off" disables it. |
| 14:19.19 | nettie | hi guys, I'm wondering is there's a way to "synchronize" the caller audible ringing tones with the actual network ones? As soon as I finish t odial a number using a sip device I can hear the ringing tone way before the phone I'm calling starts ringing. This is pretty annoying I rather prefer having silence during signalling processing before the actual destination phone starts rining. Anyone know if this behaviour is configurable? Thanks |
| 14:20.43 | kaldemar | nettie: it is your sip device that generates the tone for you. if it starts to generate it before it gets progress information, there's probably little you can do. |
| 14:20.53 | WIMPy | That should be the normal way. Is it about local phones or where ere you calling to? |
| 14:23.29 | nettie | well I'm calling from a polycom sip phone to a mobile phone via a BRI card |
| 14:24.30 | RadJackson | kaldemar can i copy what sip set debug says ? |
| 14:25.17 | RadJackson | pastebin [DOT] com/dvf5xbGJ |
| 14:25.18 | WIMPy | nettie: In that case you probaly want progressinband=yes |
| 14:26.16 | kaldemar | RadJackson: you don't have to garble links that you paste here. :) |
| 14:26.34 | RadJackson | Ok sorry |
| 14:26.51 | RadJackson | http://pastebin.com/dvf5xbGJ |
| 14:27.40 | WIMPy | RadJackson: Noone answered the call within the time you cave them. |
| 14:27.44 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:27.45 | kaldemar | paste the debug for the whole call, not just a single message. |
| 14:28.37 | igcewieling | nettie: remove the "r" option from Dial if you use it |
| 14:30.22 | RadJackson | WIMPy actually it's a little bit complicated, we place two .call files into outgoing folder, in order to leave a message directly into the voice mail |
| 14:30.42 | RadJackson | both .call files are executed at the same time , the first one hold the line busy , second gets directly into voice mail and leave a message |
| 14:30.53 | RadJackson | it actually works , we have been testing 100% working |
| 14:31.11 | WIMPy | Why do you need two files? |
| 14:32.38 | RadJackson | two call files ,calling same number, both has a WaitTime: 5 , when the first calls, after 1 second the second file does the same , the 1st stops , and the second leaves a message in the voicemail |
| 14:32.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:32.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:33.35 | WIMPy | I think I get the idea. Interesting thing. So it's normal for the 1st call to just time out. |
| 14:34.07 | RadJackson | Yes :) we have been experimenting this for like weeks , its working perfect :) |
| 14:34.39 | RadJackson | but when do place many call files , like 50 , which mean 25 numbers to call , it starts to cancel some calls |
| 14:36.17 | WIMPy | How are you calling out? Maybe you're too fast for your provider? Or have a channel limit? |
| 14:37.37 | RadJackson | we have got access to 5 calls simultenously , we use .call files , we place two files every minute |
| 14:37.51 | RadJackson | we do not think that it is a provider limit issue |
| 14:38.36 | WIMPy | Only some more debugging can tell you. |
| 14:38.42 | Qwell | RadJackson: So, your calls are only 2.5 minutes long? |
| 14:39.16 | RadJackson | Qwell the first file holds the line busy for 2 seconds , the second file Waits for silence and leave a message on the voicemail |
| 14:39.24 | RadJackson | so both files are executed in less than 1 minute |
| 14:39.37 | RadJackson | by file i mean "file.call" |
| 14:39.38 | WIMPy | You can send the call files to the dialplan instead of calling directly. That gives you the opportunity to place some Verbose() after the dial attempt. |
| 14:43.16 | RadJackson | The first .call file contains this : |
| 14:43.36 | RadJackson | Channel: SIP/TRUNK/CustomerNumber - CallerID: XXXXXXXX - Context: MyContext - Extension: s - WaitTime: 5 |
| 14:43.39 | RadJackson | the second : |
| 14:43.51 | RadJackson | Channel: SIP/TRUNK/CustomerNumber - CallerID: XXXXXXXX - Context: MyContext2 - Extension: s - WaitTime: 5 |
| 14:44.36 | Qwell | wait, you're calling somebody, in order to block their line? |
| 14:44.45 | Qwell | That's super shady. |
| 14:44.52 | RadJackson | Nop in order to leave a message thru voicemail |
| 14:45.07 | RadJackson | the first call hold the line for 5 seconds |
| 14:45.37 | RadJackson | MyContext : exten => s,1,NoOp( Start Call ) exten => s,2,WaitForSilence(2000) exten => s,3,PlayBack(/var/spool/asterisk/sounds/${AudioFile}) exten => s,4,HangUp() |
| 14:45.54 | RadJackson | MyContext2 : exten => s,1,NoOp( Whatever extension to make line busy )- exten => s,2,HangUp() |
| 14:46.00 | RadJackson | Thats all |
| 14:46.08 | Qwell | I'm pretty sure that's illegal in the US. Just sayin'. |
| 14:46.24 | RadJackson | may be |
| 14:46.45 | RadJackson | the main purpose is to leave a message thru voicemail without bothering |
| 14:48.30 | RadJackson | What are the cases on which you get this error message : Call failed to go through, reason (3) Remote end Ringing |
| 14:48.57 | igcewieling | RadJackson: when the far end does not answer |
| 14:49.41 | WIMPy | That's what you should get for all "first" calls. |
| 14:50.18 | RadJackson | why when i test on three numbers , it works perfect? i never see that error message, but only when i do massive calls |
| 14:51.02 | Qwell | igcewieling: I'd recommend not helping here. If not for the questionable legality, for the shadiness... |
| 14:51.20 | RadJackson | we speak tech here |
| 14:51.38 | igcewieling | Qwell: noted. BTW, welcome to the Grumpy Tech Club. [TK]D-Fender and I welcome you. |
| 14:51.38 | RadJackson | i may do some experiments on my phones |
| 14:51.41 | RadJackson | is that illegal? |
| 14:51.57 | Qwell | igcewieling: Please, I'm the founder. :p |
| 14:52.16 | igcewieling | Qwell: I disagree, you are usually much less grumpy. Here is your toaster oven. |
| 14:52.41 | mjordan | wait wait wait WAIT. You get a toaster oven??? |
| 14:53.25 | igcewieling | mjordan: it is a Ellen DeGeneres reference |
| 14:53.40 | RadJackson | thanks everyone for the help , i appreciate. |
| 14:53.43 | mjordan | All I heard was "free toaster oven" |
| 14:53.53 | mjordan | likes him some toast |
| 14:53.54 | igcewieling | mjordan: all you heard was "free" 8-) |
| 14:54.02 | WIMPy | thought roaster oven was a reference to an EWSD :-) |
| 14:54.16 | [TK]D-Fender | I have a toaster. I have an oven. Why would I need a toaster oven? |
| 14:54.29 | jrose_atDigium | RadJackson: If you are interested in learning the purpose of any given log message, reading the source a bit usually, ahhhh... he left. |
| 14:54.33 | igcewieling | I just have an oven. It works fine for toast. |
| 14:54.34 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 14:54.40 | [TK]D-Fender | I also got a crock-pot as a company Christmas gift... wtf am I going to do with that? |
| 14:54.49 | mjordan | make some bitchin soup |
| 14:54.51 | Qwell | [TK]D-Fender: are you kidding? |
| 14:54.54 | Qwell | crock-pots are amazing. |
| 14:54.57 | [TK]D-Fender | Qwell, nope. |
| 14:54.59 | mjordan | what *can't* you do with crock-pots |
| 14:55.05 | Qwell | mjordan: beef jerky |
| 14:55.06 | igcewieling | [TK]D-Fender: break it then you have your own crackpot? |
| 14:55.17 | [TK]D-Fender | Qwell, Substantially less amazing when everything I cook is fast & easy .... and you're cooking for 1. |
| 14:55.18 | mjordan | igcewieling: well played |
| 14:55.31 | [TK]D-Fender | igcewieling, I already have crack-pots HERE :p |
| 14:55.42 | Qwell | [TK]D-Fender: what's easier than throwing a huge piece of meat and some water in, and ignoring it for a day? |
| 14:55.47 | igcewieling | tends to follow the George Carlin ideas on "stuff" |
| 14:56.20 | Qwell | also, your house smells delicious for like 6 hours. |
| 14:56.22 | [TK]D-Fender | Qwell, Not having to package and reheat the tons of leftovers validating the process requires. And I can't grill on it. |
| 14:57.34 | [TK]D-Fender | Qwell, it's still in it's box since December :) |
| 14:57.38 | [TK]D-Fender | its* |
| 14:58.41 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:59.45 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 15:00.11 | *** join/#asterisk butthurtface (~Butthurtf@38.122.108.2) |
| 15:04.28 | igcewieling | Stuff: http://www.youtube.com/watch?v=MvgN5gCuLac |
| 15:05.37 | *** join/#asterisk navaismo (~navaismo@189.241.51.199) |
| 15:08.28 | igcewieling | [TK]D-Fender: your best option is....re-gifting! |
| 15:09.37 | *** join/#asterisk sheebacatfood (~ulaw@c-174-53-131-243.hsd1.mn.comcast.net) |
| 15:10.16 | caterwaul | good morning |
| 15:10.36 | caterwaul | this isn't an asterisk question but voip-related. have any of you ever compiled your own version of csipsimple? |
| 15:10.37 | caterwaul | http://code.google.com/p/csipsimple/wiki/HowToBuild |
| 15:10.39 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-30.usfamily.net) |
| 15:13.57 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 15:15.33 | butthurtface | 700 isn't enough for a weekend hooker |
| 15:15.46 | butthurtface | generally looking at $1200 for an overnighter with a mid-grade escort. |
| 15:17.05 | caterwaul | Why don't you considering holstering it for a change you primitive lizard brain? |
| 15:17.12 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 15:17.24 | caterwaul | consider |
| 15:22.59 | itgrl | How is everyone this morning? |
| 15:23.47 | caterwaul | on the cusp of making video tutorials from start to finish for asterisk + google voice + android |
| 15:23.55 | caterwaul | tired of misinformation and bullshit |
| 15:24.05 | caterwaul | it took me WEEKS to get my server set up |
| 15:26.20 | itgrl | just make sure everything is properly licensed if talking about it in here. |
| 15:26.43 | caterwaul | here we go again with the truman show |
| 15:27.06 | caterwaul | i like open communication. you know--the nice thing about the internet? I don't like rules and limitations yet it's everywhere you go. |
| 15:27.42 | [TK]D-Fender | caterwaul, If there weren't rules I'd have left a trail of bodies that would make Pol Pot blush. |
| 15:27.53 | caterwaul | my girlfriend's Cambodian |
| 15:28.05 | [TK]D-Fender | caterwaul, And there are rules. Welcome to the real world. |
| 15:28.22 | caterwaul | you can't talk about pol pot because it offends me. *feigns indignance* |
| 15:28.34 | caterwaul | so if there are rules |
| 15:28.41 | caterwaul | let's get on skype, jitsi, mumble, or ventrilo. |
| 15:28.46 | caterwaul | voice is better, right? |
| 15:29.10 | caterwaul | hey fender |
| 15:29.26 | caterwaul | I uploaded a video today of me using my cell phone in my car with the g.729 codec. do you have a problem with that? |
| 15:29.34 | navaismo | seriously? Again... Don't feed the troll! |
| 15:29.42 | caterwaul | I mean what I say, navaismo |
| 15:29.57 | caterwaul | [23:49] <Andrew__> remember caterwaul? |
| 15:29.59 | caterwaul | [23:50] <Andrew__> I just talkied with him for 36 mins |
| 15:30.01 | caterwaul | [23:50] <Andrew__> he is NOT a troll |
| 15:30.04 | caterwaul | [23:50] <Andrew__> he is super passionate and awesome |
| 15:30.16 | caterwaul | don't ever use that disingenuous word again |
| 15:30.41 | caterwaul | http://www.youtube.com/watch?v=a_EOe8XNuVs&hd=1 |
| 15:30.47 | caterwaul | there's the video of me using the g.729 codec. |
| 15:30.48 | [TK]D-Fender | navaismo, Indeed you should not call him a troll... |
| 15:30.50 | caterwaul | want to make something out of it? |
| 15:31.24 | *** mode/#asterisk [+b *!*nirv@*2001 19f0:1619:c9::c805:4072] by Qwell |
| 15:31.24 | *** kick/#asterisk [caterwaul!~north@pdpc/sponsor/digium/Qwell] by Qwell (No, you should call him a jackass, instead.) |
| 15:31.38 | navaismo | haha |
| 15:33.04 | igcewieling | hugs his /ignore button |
| 15:33.41 | [TK]D-Fender | navaismo, He is not a troll. A troll does what he does just to get a rise out of you. He is actually that much of an ignorant rude self-entitled delusional asshole. "It's not a lie .... if you believe it" - George Costanza |
| 15:34.05 | butthurtface | Wow someone pissed off the herd. |
| 15:34.14 | Greenlight | lol he almost had it a full 30 mins before getting banned |
| 15:34.23 | navaismo | [TK]D-Fender, the buzz lightyear quote apply to him |
| 15:34.41 | [TK]D-Fender | navaismo, Which? |
| 15:34.42 | *** join/#asterisk evilman_home (kvirc@78-106-161-124.broadband.corbina.ru) |
| 15:34.48 | butthurtface | To infinity and beyond? |
| 15:34.52 | igcewieling | butthurtface: he has not spammed you via /msg yet? |
| 15:34.57 | Qwell | [TK]D-Fender: You are a sad, strange little man, and you have my pity. |
| 15:34.57 | butthurtface | Nope |
| 15:35.00 | navaismo | yes already does |
| 15:35.11 | navaismo | Yes that Quote Qwell |
| 15:35.18 | [TK]D-Fender | Qwell, o_O |
| 15:35.23 | Qwell | [TK]D-Fender: quote |
| 15:35.27 | Greenlight | Heh why's he /msg ing me ffs |
| 15:35.35 | Qwell | Greenlight: yeah, he does that |
| 15:35.45 | Greenlight | BUt I kept quiet till he'd gone! |
| 15:35.54 | MLNoah | I have an Asterisk 11.2.1 server running a small office PBX. The outside lines are PSTN into a Cisco SPA8800, which then connects via SIP to Asterisk. There are 2 Cisco SPA525G2 phones that are the primary handsets, and 2 additional analog DECT phones (connected via SIP ATAs, one is the same SPA8800 that handles the FXO lines, and one is a Cisco SPA112 in another building) for roaming... |
| 15:35.56 | Qwell | He's probably messaging the channel. |
| 15:35.56 | MLNoah | ...users. If an incoming call gets answered by the SPA525G2 phones, the first few seconds of the call sound really soft and/or garbled, but then the rest of the call is fine. If an incoming call gets answered by one of the DECT phones, the entire call is fine. Changing jitterbuffer settings, codec availability, and upgrading firmware on the SPA525G2s hasn't helped. Any suggestions what... |
| 15:35.57 | MLNoah | ...to try next? |
| 15:35.59 | [TK]D-Fender | Qwell, Please actually quote it when you answer my question to someone else :) especially for the context it implies! |
| 15:36.10 | Qwell | heh, sorry |
| 15:36.16 | Qwell | [TK]D-Fender: I thought about that after I sent |
| 15:36.24 | navaismo | ah you guys always make me smile |
| 15:36.39 | mjordan | MLNoah: which channels have the jitter buffer assigned to them? |
| 15:37.15 | igcewieling | I suspect he is from EFNet and travelled to the present day from 2001 |
| 15:37.19 | [TK]D-Fender | Qwell, I had thought my preceding argument was entirely literate and accurate :) |
| 15:38.10 | Qwell | igcewieling: Hopefully he won't DDoS us then. :p |
| 15:38.43 | igcewieling | What makes you think he won't? |
| 15:39.03 | [TK]D-Fender | MLNoah, "soft" would imply that the level differnce for those DECT's is probably a better initial match for the EC on the 8800 to manage so it homes in faster |
| 15:39.50 | MLNoah | mjordan: my original config was no jitterbuffer on asterisk's SIP channels. Also, Asterisk is remaining in the call path (directmedia = no), and the only audio problem is the leg from the SIP phone to the external lin. |
| 15:40.04 | butthurtface | [TK]D-Fender has always been helpful and nice towards me. No problems with him :) |
| 15:41.09 | MLNoah | it's kind of frustrating because it seems to be variable -- it's ranged anywhere from the user behind the SIP phone being completely inaudible to the external caller, to them sounding choppy & robotic (kind of like you get when you have a bad bluetooth connection to your cellphone) |
| 15:41.22 | MLNoah | and apparently on some calls it works reasonably well from the start. |
| 15:43.57 | mjordan | MLNoah: how are you applying the jitter buffer? Through the configuration, or through func_jitterbuffer? |
| 15:44.03 | mjordan | i.e., JITTERBUFFER function |
| 15:44.07 | MLNoah | i tried through jbenable=yes in sip.conf |
| 15:44.10 | mjordan | ah ha |
| 15:44.33 | mjordan | try enabling the jitter buffer by applying the JITTERBUFFER function on the SIP channel in the dialplan |
| 15:44.51 | mjordan | or, enable it on the DAHDI channels |
| 15:44.52 | MLNoah | at the moment my inbound dial is a ring-all strategy |
| 15:44.56 | mjordan | in their configuration |
| 15:44.58 | MLNoah | no DAHDI in this box. |
| 15:45.10 | igcewieling | isn't de-jittering a function of the ENDPOINT? |
| 15:45.12 | MLNoah | SIP to the SPA8800 |
| 15:45.23 | mjordan | oic. Well, use the JITTERBUFFER function on the SIP channel instead of the configuration |
| 15:45.27 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 15:45.44 | Rahail | HI every one pleaase dont get me wrong I asked already asterisk mailling list for some reason didnt got reply |
| 15:46.28 | Rahail | I will pay for it if some one can make me a dialplan where if i have few trunks and trunk 1 call hit now i need to put delay interval not to send call this trunk for next x second or minute it need to use next trunk |
| 15:46.38 | mjordan | configuration application of jitter buffers apply them on the write side of the channel, which is rarely what you actually want. If you have a DAHDI channel bridged with a SIP channel, that means you have to set the jitter buffer (via config) on the DAHDI channel, as you need to de-jitter the jittered frames in the Asterisk core. The JITTERBUFFER function in 10+ puts the jitter buffer on the read side of the channel |
| 15:46.57 | mjordan | so you put the jitter buffer on the channel that actually has jitter, as opposed to every other channel in the system |
| 15:47.29 | navaismo | Rahail, what? |
| 15:47.46 | MLNoah | so, since i'm ringing a bunch of different endpoints, I should change the two effected endpoints to be rung via a Local channel, so I can turn on JB just for those two channels? |
| 15:47.58 | Rahail | I need to make a dilplan where that puts delay to send call to x trunk if call was process prvious |
| 15:48.18 | mjordan | MLNoah: or just put it on the SIP channel itself using the function. Then, whoever it dials doesn't matter |
| 15:48.35 | apb1963 | navaismo: He wants to load balance his trunks |
| 15:48.48 | Rahail | not loadblance |
| 15:48.48 | mjordan | If you're dialing the SIP devices, in 11 you can apply the jitterbuffer via the function using pre-dial |
| 15:49.18 | mjordan | MLNoah: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers |
| 15:49.33 | mjordan | it lets you set properties on outbound channels, so you don't have to resort to Local channels and inheritance options |
| 15:49.53 | mjordan | specifically, the lowercase 'b' option in Dial |
| 15:50.13 | navaismo | hi apb1963 |
| 15:50.24 | MLNoah | but that's going to execute on each channel. I'm using Dial(SIP/badphone1&SIP/badphone2&SIP/atathatworks1&SIP/atathatworks2,...) |
| 15:50.32 | mjordan | ah... |
| 15:50.41 | Rahail | navaismo can u o do it or apb1963 |
| 15:50.57 | mjordan | welp. You can either return if the SIP device being dialed isn't the one you want (string manipulation in the dialplan), or you could do what you said and use Local channels |
| 15:51.06 | apb1963 | Rahail: You're saying that if trunk 1 is hit, then to delay it's use for X seconds, and to use trunk 2 while trunk 1 is delayed. Yes? |
| 15:51.10 | navaismo | Rahail, well since i dont understan what you want then no... :) |
| 15:51.10 | mjordan | I'd probably do string manipulation myself, simply because Local channels add complexity (tm) |
| 15:51.13 | apb1963 | Hello navaismo |
| 15:51.20 | apb1963 | how're you? |
| 15:51.22 | MLNoah | right, ok. |
| 15:51.25 | Rahail | can pm you |
| 15:51.41 | mjordan | MLNoah: of course, that all assumes that it really is jitter. Wireshark would tell you that too |
| 15:51.44 | Rahail | so ic na expalin you better i dont want flood the channel with my stuff |
| 15:52.16 | navaismo | apb1963, fine just trying to sell some PICMicros |
| 15:52.29 | apb1963 | Rahail: No, I can't help you. I don't know how to do it. |
| 15:52.49 | apb1963 | Rahail: If it's that much information, then pastebin it |
| 15:52.51 | *** join/#asterisk aruntomar (~Thunderbi@49.248.154.38) |
| 15:52.59 | MLNoah | in older versions of Asterisk, I remember seeing the console output "Locally bridging" or "Remotely bridging" depending on whether it set up direct RTP. But it doesn't look like this box is doing either... how do you tell if Asterisk is in the call path with 11? |
| 15:53.06 | apb1963 | navaismo: Wish I could help you |
| 15:53.24 | navaismo | apb1963, dont worry no one cant help me, hehe |
| 15:53.25 | apb1963 | navaismo: sales is a hard business |
| 15:53.56 | apb1963 | especially computer sales... too much competition... margins are too slim. |
| 15:54.20 | igcewieling | MLNoah: 11.3: -- Locally bridging SIP/ipmax-angiuli-bay-00021e50 and SIP/level3-00021e51 |
| 15:54.32 | MLNoah | hrm. |
| 15:54.41 | igcewieling | MLNoah: what verbosity level are you running at? |
| 15:54.50 | MLNoah | i think this is officially my "i hate IT" moment of the day. verbose 3 |
| 15:54.55 | navaismo | im so fat now |
| 15:55.47 | MLNoah | even at verbose 5, all i get is -- SIP/101-00000006 answered SIP/spa8800-fxo1-00000005 with no locally or remotely bridging data |
| 15:56.12 | igcewieling | MLNoah: on my systems the next line is the bridging line |
| 15:56.32 | MLNoah | yeah, that's where i was expecting it too. |
| 15:56.40 | MLNoah | blah. i'll come back to it with more calories in my stomach. |
| 15:56.49 | MLNoah | thanks for the help, guys. |
| 15:56.51 | apb1963 | So, I'm sending a message to asterisk using the manager interface... it's returning: Response: Error Message: Extension does not exist. |
| 15:57.11 | apb1963 | But... it does exist. So.... how do I got about finding the problem? |
| 15:57.45 | igcewieling | apb1963: what CONTEXT is the extension in and what context are you specifying in your AMI request? |
| 15:57.47 | navaismo | show us the entire output |
| 15:58.31 | [TK]D-Fender | Actaull show us the entire INPUT |
| 15:59.19 | apb1963 | navaismo: http://ix.io/5sG |
| 15:59.36 | apb1963 | igcewieling: hang on |
| 15:59.48 | apb1963 | [TK]D-Fender: you too :) |
| 16:00.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:00.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:01.31 | nettie | sorry guys I went away. I solved with progressinband=yes and prematuremedia=no -- thanks! |
| 16:02.12 | apb1963 | igcewieling: http://ix.io/5sH |
| 16:02.47 | Qwell | XXX1? |
| 16:02.49 | igcewieling | apb1963: that is half the info I asked about? |
| 16:03.03 | igcewieling | Qwell: good catch |
| 16:03.13 | apb1963 | igcewieling: You're right... I only read the first half of the question |
| 16:03.24 | igcewieling | typical freepbx mixing patterns |
| 16:03.26 | apb1963 | I don't understand the "catch" |
| 16:03.45 | Qwell | Your pattern and priorities don't make sense. |
| 16:04.22 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 16:04.50 | apb1963 | ok, I'm not seeing it |
| 16:04.53 | apb1963 | why? |
| 16:04.55 | navaismo | first line |
| 16:05.03 | apb1963 | yes? |
| 16:05.04 | navaismo | end with 1 |
| 16:05.13 | apb1963 | so? |
| 16:05.15 | Qwell | apb1963: What do you think Asterisk will think the priority is, on line 3 of your pastebin? |
| 16:05.22 | przerull | hello. so I have to answer my inbound leg immediatly upon connection. Is there a way to generate ringing that stops when we recieve progress from the outbound leg? |
| 16:06.04 | navaismo | apb1963, 860XXX1 then 860XXXX |
| 16:06.07 | igcewieling | navaismo: that will work but it is a HORRIBLE thing to do and will only work for extensions ending in 1 |
| 16:06.10 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:06.15 | navaismo | yes yes |
| 16:06.16 | [TK]D-Fender | apb1963, Where is the complete AMI request? |
| 16:06.28 | igcewieling | navaismo: typical freepbx style perversion of Asterisk's pattern matching |
| 16:06.43 | navaismo | hahaha |
| 16:07.25 | apb1963 | [TK]D-Fender: Are you in disagreement with what they;'re saying here? |
| 16:07.43 | navaismo | no he want to see the AMI request |
| 16:07.45 | igcewieling | apb1963: please post your AMI request |
| 16:07.45 | Qwell | [TK]D-Fender: He can't agree or disagree. He hasn't seen the failure. |
| 16:07.47 | navaismo | and also I |
| 16:07.52 | navaismo | and igcewieling |
| 16:08.00 | apb1963 | ok.. that will take a few minutes to put together |
| 16:08.07 | [TK]D-Fender | apb1963, I'm saying I don't care how stupid the dialplan you're showing is when I can't prove your request even looks in its general vicinity. |
| 16:08.14 | apb1963 | ok |
| 16:08.19 | igcewieling | apb1963: do NOT change the telephone numbers |
| 16:08.33 | navaismo | apb1963, i want a sandwich |
| 16:08.35 | apb1963 | I won't |
| 16:08.42 | apb1963 | navaismo: you're getting fat |
| 16:08.56 | navaismo | im already fat like 100Kg |
| 16:08.58 | apb1963 | give me a few minutes please |
| 16:11.39 | apb1963 | http://ix.io/5sI |
| 16:11.57 | Qwell | There is no priority 1 for that extension. |
| 16:12.03 | Qwell | also, Default? |
| 16:13.30 | apb1963 | This was written in the 1800's... I'm trying to fix it for the current version. |
| 16:13.56 | apb1963 | very good catch... I've stared at that for hours |
| 16:14.17 | apb1963 | although I'm still scratching my head over your priority comment |
| 16:14.34 | igcewieling | apb1963: "(11:57:45 AM) igcewieling: apb1963: what CONTEXT is the extension in and what context are you specifying in your AMI request?" |
| 16:14.48 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 16:14.59 | apb1963 | good call igcewieling |
| 16:15.06 | igcewieling | apb1963: but it still won't work |
| 16:15.15 | apb1963 | because the dialplan isn't right |
| 16:15.21 | igcewieling | correct. |
| 16:15.41 | apb1963 | hmm |
| 16:15.42 | navaismo | apb1963, 860XXX1 only will work for extensions 860[0-9][0-9][0-9]1 |
| 16:15.51 | igcewieling | the extension you specify8600052 will NEVER match 860XXX1 |
| 16:15.58 | apb1963 | right |
| 16:16.16 | apb1963 | so let me think about what I want |
| 16:16.25 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:16.43 | apb1963 | Any reason I can't use use X instead of 1 there? |
| 16:17.23 | apb1963 | 860XXXX |
| 16:17.26 | apb1963 | ? |
| 16:19.39 | navaismo | hmm? You can use the X. if you want [0-9] matching at the end too |
| 16:22.17 | igcewieling | apb1963: putting X there is the Correct Thing to Do |
| 16:22.42 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 16:22.47 | gnudna | Hi guys |
| 16:23.16 | apb1963 | thank you :) what about the priority? |
| 16:23.39 | apb1963 | I presume changing to X will fix the priority issue? |
| 16:23.50 | gnudna | weird issue when i call an extension hang up and call back within 2 -3 seconds it seems to go straight to voicemail |
| 16:24.11 | gnudna | even though the phone all have 2 lines setup minimum |
| 16:24.18 | navaismo | apb1963, yes priority numbers are ok |
| 16:24.54 | gnudna | anybody able to help or point in the right direction? |
| 16:24.56 | [TK]D-Fender | gnudna, pastebin the complete call CLI output : "core set verbose 10", "sip set debug on". |
| 16:24.59 | [TK]D-Fender | ~pb |
| 16:24.59 | 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. |
| 16:25.01 | [TK]D-Fender | ^^^ |
| 16:25.02 | apb1963 | thank you one and all :) |
| 16:25.13 | gnudna | ok Fender |
| 16:25.17 | gnudna | hold on please |
| 16:29.31 | przerull | hello. so I have to answer my inbound leg immediatly upon connection. Is there a way to generate ringing that stops when we recieve progress from the outbound leg? |
| 16:31.03 | igcewieling | przerull: have you tried Ringing() ? |
| 16:31.42 | igcewieling | of course if the far end is busy then you get ringing followed by busy. |
| 16:31.55 | przerull | i'm ok with ringing followed by busy |
| 16:32.14 | przerull | the issue is that i've already answered the inbound leg so ringing won't play anything |
| 16:32.15 | gnudna | D-Fender is there a way to isolate the output for a specific extension? |
| 16:32.29 | [TK]D-Fender | gnudna, nothing reliable |
| 16:32.47 | gnudna | ok let me see if i can get the relevant section |
| 16:33.01 | gnudna | does not seem to be writing this to log file after i enabled dbug |
| 16:33.09 | gnudna | just on the asterisk console |
| 16:33.19 | przerull | ok so i was mistaken in that sence |
| 16:33.27 | przerull | ringing will play after answer |
| 16:34.00 | przerull | but what's happening is that the ringing stops once it gets put into a meetme conference |
| 16:34.54 | przerull | now I can use the r option in the dial command that does my outbound leg |
| 16:35.30 | przerull | but then the ringing plays past the progress from the far side (breaking early media) and plays until answer |
| 16:36.16 | *** join/#asterisk polysics (~Adium@4.31.112.6) |
| 16:36.37 | polysics | hi there |
| 16:37.17 | polysics | does anyone please know if /how it is possible to set up a menu for 1.8 ConfBridge so that the admin can kick users fro ma conf using DTMF? |
| 16:37.39 | przerull | have you tried features.conf? |
| 16:38.28 | polysics | I know 10/11 ConfBridge has menus you can put in confbridge.conf but it either doesn't work on 1.8 or the syntax is different |
| 16:38.29 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 16:39.18 | polysics | but there is no documentation at all about this stuff, at least that I could find |
| 16:39.40 | mjordan | polysics: it isn't possible. ConfBridge was rewritten in 10. |
| 16:40.02 | mjordan | And there is documentation on ConfBridge on the Asterisk wiki which talks about ConfBridge in 10+ |
| 16:40.03 | polysics | mjordan: thanks, finally I find out :-) |
| 16:40.14 | mjordan | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 16:40.19 | polysics | yes, the 1.8 version seems not documented but I understand why |
| 16:40.38 | polysics | thanks, I just needed closure here |
| 16:40.53 | polysics | mjordan: btw, I am Luca from the Adhearsion crew :-) |
| 16:41.01 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 16:41.01 | mjordan | ah, hey there :-) |
| 16:42.02 | mjordan | You might consider ConfBridge in 1.8 and prior versions as a proof of concept of the Bridging Framework. 10+ was taking ConfBridge to a fully featured application. |
| 16:42.37 | polysics | we might be upgrading that project to 11 for other reasons, might as well chalk this up on the list too |
| 16:43.43 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 16:43.50 | polysics | the main issue here was not the feature being available or not, but finding out if it was :-) |
| 16:44.19 | *** part/#asterisk przerull (~philip@50.56.205.232) |
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| 16:54.32 | pancho_jay | Hi |
| 16:54.46 | pancho_jay | I need some help to setup my dialplan |
| 16:55.03 | pancho_jay | someone speaks spanish? |
| 17:00.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.104) |
| 17:04.09 | igcewieling | Has anyone seen this sort of message when trying to forward a VM to another mailbox from inside VoiceMailMain. The destination mailbox is NOT full. [2013-05-03 10:43:32] NOTICE[3776] app_voicemail.c: Mailbox '2114' is full with capacity of 100, prompting for another extension. |
| 17:04.25 | igcewieling | 2114 is the dest mailbox |
| 17:05.42 | igcewieling | I think I see the issue, the Old folder has 99 messages. I would expect the fwd'd message to be put in the new messages (INBOX) folder |
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| 17:10.57 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.static.tpgi.com.au) |
| 17:15.39 | blitzrage | damnit, what the hell is the command to hangup a channel in 1.4? :) |
| 17:16.18 | igcewieling | blitzrage: soft<tab> maybe? |
| 17:16.25 | blitzrage | oh right |
| 17:16.28 | blitzrage | it's under soft, thanks |
| 17:16.30 | igcewieling | blitzrage: the memory is the 2nd thing to go. |
| 17:16.55 | blitzrage | I know... it's been literally several years since I've used 1.4 |
| 17:18.52 | blitzrage | :) |
| 17:19.05 | navaismo | Yo pancho |
| 17:20.44 | *** mode/#asterisk [+b *!*@c-174-53-131-243.hsd1.mn.comcast.net] by Qwell |
| 17:20.47 | *** kick/#asterisk [sheebacatfood!~north@pdpc/sponsor/digium/Qwell] by Qwell (sheebacatfood) |
| 17:21.01 | navaismo | Cual es el problema pancho_jay ? |
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| 18:04.05 | gnudna | [TK]D-Fender, here is the pastebin sorry for the delay but got dragged into another issue. http://pastebin.com/KUWsMCTf |
| 18:04.52 | carrar | FYI: It's issue free Friday |
| 18:05.17 | gnudna | as stated seems calls go to voicemail if i call the same extension back to back. |
| 18:05.40 | gnudna | phone in question is setup for 2 active lines |
| 18:05.52 | gnudna | btw this seems to happen on all the xtensions i tested |
| 18:06.04 | gnudna | some have 3 or 4 lines |
| 18:10.45 | igcewieling | gnudna: using IAX2 complicated things. I see only one Dial line and I see the call ringing the SIP phone, then I see then caller hangup. |
| 18:13.41 | *** part/#asterisk polysics (~Adium@4.31.112.6) |
| 18:15.12 | gnudna | agreed but why the direct voicemail on second call |
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| 18:17.12 | pancho_jay | navaismo, perdon, estaba comiendo un asado ;) |
| 18:17.15 | pancho_jay | como estas? |
| 18:19.30 | navaismo | uh provecho! bien gracias cual era el problema con el dialplan |
| 18:20.03 | igcewieling | gnudna: what line in your paste shows the call going to voicemail? |
| 18:20.42 | pancho_jay | navaismo, la cosa es así.... tengo un proveedor SIP donde termino las llamadas, cuando se agota el saldo el me devuelve error SIP 503 circuit busy |
| 18:21.03 | navaismo | ok... |
| 18:21.07 | pancho_jay | la idea es que mis clientes reciban un codigo de error que les permita reenrutar los llamados |
| 18:21.15 | pancho_jay | mis clientes son otras telcos |
| 18:21.19 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:21.27 | pancho_jay | navaismo, lo ves viable? |
| 18:22.20 | navaismo | me suena a que necesitas un proxy sip |
| 18:24.03 | pancho_jay | navaismo, mmm como sería eso? |
| 18:25.42 | navaismo | necesitas algo como kamalio par que maneje las respuestas, nunca lo he usado pero aqui y en google he visto mucho eso para rutear en base a los mensajes sip |
| 18:26.08 | gnudna | igcewieling, im not seeing them in that pastebin |
| 18:26.20 | igcewieling | gnudna: then you have a bad pastebin |
| 18:26.58 | igcewieling | You can disable iax and sip debug if you want. We may need the information, but right now I just want to see the CLI showing the problem call |
| 18:27.15 | pancho_jay | navaismo, se te ocurre como podria hacer un script para ver si el trunk esta caido o esta rechazando las llamadas? |
| 18:27.19 | igcewieling | you have until our tech gets to the customer location then I have to concentrate on customer issues |
| 18:27.37 | gnudna | igcewieling, thanks |
| 18:27.52 | gnudna | i will try to reproduce it again |
| 18:29.37 | navaismo | pancho_jay, si tienes el Qualify activado podrias checar el status y en base a eso tomar otra linea, en el dial plan con el HANGUPCAUSE o DIALSTATUS tambien se puede, dependiendo de la version de tu asterisk puedes saber la respuesta SIP con HANGUPCAUSE o HASH(SIP_CAUSE) |
| 18:29.46 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:30.01 | pancho_jay | navaismo, buena idea |
| 18:30.08 | pancho_jay | tengo asterisk 1.8 |
| 18:30.15 | pancho_jay | mis clientes y proveedor usan SIP |
| 18:30.23 | navaismo | ese usa HASH(SIP_CAUSE |
| 18:30.42 | gnudna | igcewieling, http://pastebin.com/h4Wyq8kY |
| 18:31.01 | gnudna | i made 2 calls the second goes to voicemail after i hangup on the first one |
| 18:31.14 | gnudna | so i call and it rings i hang up and call again voicemail |
| 18:32.27 | igcewieling | I still don't see the Asterisk voicemail application being run |
| 18:33.34 | gnudna | is verbose 10 enough? |
| 18:33.43 | pancho_jay | navaismo, gracias! |
| 18:33.48 | pancho_jay | por que usaria HASH? |
| 18:34.02 | gnudna | cause when i call i see the info scrolling but on my 2nd attempt i do not see anything in the console |
| 18:34.08 | gnudna | just what i pasted for you |
| 18:35.25 | gnudna | could this in sip.conf be causing the issue? |
| 18:35.28 | gnudna | ignoresdpversion=yes |
| 18:35.51 | navaismo | pancho_jay, lo que he leido es la forma de obtener el valor---> NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})}) |
| 18:35.59 | gnudna | this seems to fix callers on hold not getting dropped but im wondering if this causes the issue currently having? |
| 18:38.51 | gnudna | actually to correct the ignoresdpversion=yes supposedly fixes the caller not getting lost when on hold |
| 18:39.00 | gnudna | be ^ |
| 18:40.16 | gnudna | igcewieling, the phone does not even ring it just goes to voicemail as if it was not even available but i am able to use it to make outbound calls and dial internal extensions |
| 18:42.42 | gnudna | is there even a point to using iax2 when we are using a sip provider? |
| 18:43.05 | gnudna | im trying to understand what needs to be cleaned up on this thing. |
| 18:45.31 | igcewieling | gnudna: How can it go to Asterisk's voicemail if there is no dialplan line being executed to run the Asterisk Voicemail application? |
| 18:45.58 | gnudna | and yet it is doing exactly that |
| 18:46.08 | igcewieling | gnudna: not according to your pastebin |
| 18:46.22 | igcewieling | maybe it is going to a provider's voicemail box? |
| 18:46.24 | pancho_jay | navaismo, ok, gracias... brb. justo me llamaron para una reunion |
| 18:46.40 | gnudna | igcewieling, i though of that but i get the voicemail i leave |
| 18:47.10 | gnudna | i get an email telling me i have voicemail and then i dial in and listen to it |
| 18:47.16 | igcewieling | gnudna: what you are claiming is not supported by your pastebin of the CLI. Nothing I can do. I wish you the best of luck. |
| 18:47.37 | gnudna | :( |
| 18:47.39 | navaismo | de nada |
| 18:48.01 | gnudna | i guess nothing but looking at the actual configs can shed some light on this issue |
| 18:48.19 | gnudna | i inherited this system from previous admin |
| 18:48.48 | gnudna | i have zero asterisk knowledge except what im learning as i go. |
| 18:48.54 | gnudna | thank you anyways |
| 18:50.20 | gnudna | might be better of starting from scratch |
| 19:01.51 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
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| 19:54.08 | xcom | gnudna: You can all ways hire a proffessional. |
| 19:56.01 | gnudna | xcom trust me i have though of it |
| 19:56.25 | gnudna | but im also trying to learn asterisk cause its pretty cool and well i will have to support this system at some point ;) |
| 19:57.44 | *** join/#asterisk FonJockey (~FonJockey@173-125-98-251.pools.spcsdns.net) |
| 19:57.52 | xcom | nothing wrong with that :) |
| 19:58.00 | xcom | good luck. |
| 19:58.15 | gnudna | i would love some hand holding though ;) to have some things explained |
| 19:58.18 | gnudna | lol |
| 19:58.58 | xcom | dont we all. google can hold your hand and more. (No not that you nasty!) |
| 19:59.11 | *** part/#asterisk pancho_jay (~pancho_ja@220-121-17-190.fibertel.com.ar) |
| 19:59.29 | xcom | this place is a good start |
| 19:59.31 | xcom | :) |
| 19:59.41 | xcom | I cam here 2yrs ago. |
| 19:59.53 | xcom | Have not been able to leave. |
| 19:59.56 | gnudna | irc has always been my friend |
| 20:00.14 | gnudna | but the #asterisk room is just the last month on and off |
| 20:01.41 | igcewieling | ~book |
| 20:01.42 | 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 |
| 20:03.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:03.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 20:05.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-165.dedicated.allstream.net) |
| 20:07.24 | gnudna | thanks for the book reference |
| 20:07.40 | gnudna | was looking for a good recommendation |
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| 20:18.29 | xcom | Thats the best book EVER |
| 20:18.49 | gnudna | just got the pdf |
| 20:19.07 | gnudna | some light reading for the weekend ;) |
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| 21:03.36 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 21:05.02 | j-fish | I have a yealink phone,is it somehow possible to answer a call through some command?instead of actually pushing a button |
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| 21:13.23 | igcewieling | j-fish: Ask Yealink |
| 21:16.32 | sp3 | stupid question, how do I remove loopback mode from SPAN on TE420? |
| 21:16.56 | sp3 | dahdi_tool gives only option to set loopback |
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| 21:18.08 | *** part/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 21:21.30 | sp3 | guess I can reload the module, but I will reset all the channels... |
| 21:23.31 | *** join/#asterisk madhatt (~madhatt@23.31.65.29) |
| 21:23.47 | *** part/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 21:23.58 | madhatt | hey all, I'm looking for a cheap but good SIP trunk provider. Hopefully pay per minute if possible, can you folks suggest anything? |
| 21:24.09 | igcewieling | I use Vitelity |
| 21:24.27 | madhatt | also, for those who heard my story yesterday about Snom phones using google analytics, Snom replied and asked that I direct these questions to marketing…. argh |
| 21:24.41 | madhatt | thanks igcewieling I'll look into that |
| 21:27.38 | ChrisInSydney | madhatt. I'll havea chat to the Snom dude I deal with |
| 21:28.13 | madhatt | right on, I really don't feel like contacting them but if you read my post on my website you'll see it seems pretty clear they are gathering some type of data from it's users :( |
| 21:29.14 | ChrisInSydney | there is an ad thingy on the main page. AFAIK you can turn it off. There is also Snom Active for auto provisioning |
| 21:30.17 | ChrisInSydney | woohoo Asterisk 11 is alive |
| 21:30.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:30.47 | madhatt | i'm on a snom 300, you should see google code in that model, I forget what firmware |
| 21:30.54 | ChrisInSydney | ./configure --libdir=/usr/lib64 |
| 21:31.58 | ChrisInSydney | madhatt. Sounds a bit wierd. Have you run a wireshark / packet trace on it. You can probably pcap off the phone its self |
| 21:33.59 | ChrisInSydney | there is also a wierd TCP port on them thats open. I've never worked out what they use it for |
| 21:37.01 | madhatt | yeah, I did pcap it and I show that my laptop (after accessing the site) connects to IP 74.125.225.196 which I "believe" is a google analytics IP |
| 21:39.11 | ChrisInSydney | whats the connect string ?? |
| 21:39.22 | ChrisInSydney | http://??? |
| 21:39.30 | madhatt | not sure, will have to recapture, one moment |
| 21:40.32 | ChrisInSydney | cool |
| 21:41.07 | *** join/#asterisk FonJockey (~FonJockey@70-11-8-207.pools.spcsdns.net) |
| 21:42.46 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 21:43.27 | *** part/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 21:44.01 | madhatt | Ok, so the first analytics software - here is the full request URI - Full request URI: http://test.wiredminds.de/track/count.js |
| 21:44.22 | madhatt | second analytics program - Full request URI: http://www.google-analytics.com/ga.js |
| 21:44.41 | ChrisInSydney | spyphones !!! |
| 21:45.50 | madhatt | here is what I show if I follow the google TCP stream |
| 21:45.50 | madhatt | GET /ga.js HTTP/1.1 |
| 21:45.51 | madhatt | Host: www.google-analytics.com |
| 21:45.51 | madhatt | Connection: keep-alive |
| 21:45.51 | madhatt | Cache-Control: max-age=0 |
| 21:45.51 | madhatt | Accept: */* |
| 21:45.51 | ChrisInSydney | Is this when you access the web interface ?? |
| 21:46.02 | *** join/#asterisk madhatt (~madhatt@23.31.65.29) |
| 21:46.11 | ChrisInSydney | patebin madhatt |
| 21:46.15 | madhatt | sors |
| 21:46.20 | ChrisInSydney | cool |
| 21:46.30 | ChrisInSydney | duck liver patebin ;-) |
| 21:46.43 | madhatt | k - http://pastebin.com/dN3h6MWT |
| 21:46.45 | ChrisInSydney | I just looked at the first one |
| 21:46.49 | ChrisInSydney | cool |
| 21:47.02 | ChrisInSydney | is that when you browse the web interface ?? |
| 21:47.07 | madhatt | of the phone yes |
| 21:47.20 | ChrisInSydney | thats the add in the top of the home page |
| 21:47.33 | ChrisInSydney | I think you can turn that off |
| 21:47.56 | madhatt | umm… I'm not sure, I see that they ARE using google analytics and even got their UA # out of the code |
| 21:48.10 | madhatt | but I'm not a pro so I'm unsure |
| 21:49.04 | ChrisInSydney | home->preferences->Advertisment: on|off <--- use this one |
| 21:49.15 | madhatt | ok, I'll do that now and refresh and see what's the up |
| 21:49.20 | ChrisInSydney | http://wiki.snom.com/wiki/index.php/Settings/advertisement |
| 21:49.27 | ChrisInSydney | my guess |
| 21:50.16 | madhatt | that's it. Once I turn that off, not only does the banner disappear but my browser is no longer showing the analytics code… still… kinda shady in my opinion |
| 21:50.56 | ChrisInSydney | yes and no |
| 21:51.03 | ChrisInSydney | by no, I mean yes |
| 21:51.06 | *** join/#asterisk lvlinux (~n1gg@c-50-142-161-228.hsd1.tn.comcast.net) |
| 21:51.26 | ChrisInSydney | but its their phone firmware, I guess they can do with it what they want |
| 21:51.52 | ChrisInSydney | if you dont like it, buy something else. |
| 21:52.02 | ChrisInSydney | trouble is, it limits your choices |
| 21:52.16 | ChrisInSydney | Yealink Yea right ! |
| 21:53.07 | ChrisInSydney | if you have enough handsetsm you can manage them via Snom Active and set myour default features up |
| 21:54.01 | ChrisInSydney | they aren't uploading configurations or SIP authentication hashes |
| 21:54.07 | madhatt | right on… well I'll stop beating that dead horse! I really like Snom phones though.. |
| 21:54.45 | igcewieling | madhatt: you should have changed it to your own Google Ads account 8-) |
| 21:55.00 | ChrisInSydney | best ones for hacking and fiddling with. Lots and lots of options. |
| 21:56.54 | ChrisInSydney | BTW, the Voip USers Conference bridge is stil up. |
| 21:57.12 | ChrisInSydney | dial(SIP/200901@login.zipdx.com) |
| 21:57.22 | ChrisInSydney | g722, 711a 711u support |
| 21:57.38 | ChrisInSydney | call in sand say Hi. Im the only one here though |
| 21:58.58 | *** join/#asterisk xcom (~wtf@pdpc/supporter/professional/seri) |
| 22:12.54 | navaismo | why sending dtmf via call file to sip device, when the device answer cant hear the dtmf but sending a message via playback it work? |
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| 22:37.32 | *** join/#asterisk FonJockey (~FonJockey@107.31.253.245) |
| 22:42.14 | WIMPy | gets a timeout trying to reach issues.asterisk.org. |
| 22:42.16 | ChrisInSydney | BTW, the Voip USers Conference bridge is stil up. dial(SIP/200901@login.zipdx.com) g722, 711a 711u support |
| 22:42.37 | ChrisInSydney | WIMPy: Hey |
| 22:43.20 | ChrisInSydney | exten => 882,1,dial(SIP/200901@login.zipdx.com) |
| 22:44.12 | WIMPy | Don't mention conferences. That makes me think about a part of my to-do list I had displaced. |
| 22:44.34 | ChrisInSydney | The VUC bridge is still up |
| 22:45.08 | ChrisInSydney | I'm it. Everyone else has either gone to bed (EU or gone to the pub US) |
| 22:45.42 | ChrisInSydney | so what are you up to ?? |
| 22:46.24 | Freeaqingme | ChrisInSydney, pubs in the eu are still open ;) |
| 22:46.33 | WIMPy | Just cam bach from a BBQ. |
| 22:46.39 | ChrisInSydney | cool |
| 22:46.43 | WIMPy | Freeaqingme: Definitely. |
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| 23:05.44 | atan | Can you use Skype Connect to get a SIP profile for inbound/outbound Skype calls (not PSTN gateway) for a set Skype username? |
| 23:08.09 | WIMPy | Wasn't Skupe Connect Skype without Skype? But their sales team should be able to tell you. |
| 23:15.28 | ChrisInSydney | VUC bridge is still up and my Samba still isnt working across the WAN |
| 23:17.17 | ChrisInSydney | Samba is working |
| 23:40.05 | igcewieling | atan: it is seldom worth your time to try connecting to services which actively try to prevent you from connecting to them -- like Skype |
| 23:41.09 | igcewieling | For example, Skype either yanked the license Digium had or refused to renew it. http://www.digium.com/en/products/software/skype-for-asterisk |
| 23:41.39 | eirirs | sounds like microsoft was behind it |
| 23:41.51 | igcewieling | Why reward a company who are jerks? |