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00:34.31 | supersoaker | So i figured out my issue btw |
00:36.28 | supersoaker | the dial app had a gosub routine and during dialing its transfered the call to the local channel which invokes a second dial command with the same gosub routine |
00:37.10 | supersoaker | so, aprently, when dial is redirected it keeps the arguments from before? So i had stacked two call screening menues on top of eachother |
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02:43.56 | mfauzir | hi all, need help please, i have installed asterisk+dahdi + TDM410P (2 FXS, 2 FXO) |
02:46.24 | WIMPy | The average psychic capabilities in here are low. So you will have to tell what you need help with. |
02:47.13 | mfauzir | one of my FXO connect to CO (Telco line), if i try to make outbound call from the extension phone , the called party has answered but the extension (calling party) always ringing (not hear sound or something from called party) |
02:49.25 | mfauzir | so, its like asterisk can not send answerd signal to extension phone |
02:49.36 | WIMPy | What about incomming calls? |
02:50.02 | WIMPy | Usually that comes right after dialling on pots, as there's no further indication. |
02:50.24 | mfauzir | incoming call, i don't test it yet. |
02:50.26 | WIMPy | But I'm not that much in to the analog stuff. |
02:51.55 | mfauzir | should i try to make incoming call first ? |
02:52.26 | WIMPy | It's always good to collect evidence. |
02:53.04 | mfauzir | ok wait.. |
02:57.45 | mfauzir | well, there is no problem with incoming call |
02:59.29 | WIMPy | So at least your hardware must be ok. |
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03:24.53 | rumpler | Hey guys, quick q; I'm reading up on OpenWRT/DD-WRT, and keep reading about people that are setting up Asterisk on their routers to make VoIP calls. I'm just wondering what the reasons/benefits behind it are (aside from it just being fun to play around with your own PBX), most voip phones can just connect to any arbitrary SIP server anyway, so why funnel everything through a router/asterisk server? |
03:25.40 | WIMPy | The same reason people use PBXs. |
03:27.44 | rumpler | WIMPy: Could you elaborate? In my (limited) experience, a PBX is essentially a router for voip calls, why bother with it when there are only one or two phones on the network, and they connect to an external PBX anyway. My best guess is that it's for funsies, but it's a project that I see come up over and over again, so maybe I'm missing some of the benefits |
03:28.17 | WIMPy | Well, in that case there is no point. |
03:28.49 | WIMPy | But you you only got an ITSP or a classic line and multiple phones, it does. |
03:29.01 | WIMPy | .. make sense, that is. |
03:30.10 | rumpler | As in, Asterisk acts as a mediator to route calls from multiple phones to a single itsp? I guess I can see the benefit there, thanks. |
03:30.34 | WIMPy | As I said: Just as with a PBX. |
03:30.50 | rumpler | thanks |
03:30.53 | WIMPy | Off course Asterisk is better at IVR stuff, if you need that. |
03:32.49 | rumpler | WIMPy: Yeah, IVR was the one real application that I could think of, but who needs IVR on a home network? (aside from tinkering, but I'm asking in more real-world terms) |
03:33.14 | WIMPy | I use it for a CAPTCHA. |
03:34.19 | rumpler | haha, for unknown numbers to have to solve before it rings your phone? (to kill bots?) |
03:34.32 | WIMPy | Exactely. |
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06:53.21 | babak | Hi, is there a variable |
06:53.38 | babak | i know call come from which span ? |
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07:45.57 | pryshebliad | Hello! |
07:46.16 | pryshebliad | Why you don't making debian packages? |
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07:46.35 | eirirs_ | pryshebliad: Why don't you? |
07:46.49 | pryshebliad | eirirs_: I can't |
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08:32.32 | vNistelroot | hi people |
08:32.43 | vNistelroot | any experience clustering asterisk with drbd? |
08:38.53 | zpotoloom | vNistelroot: did some testing a while back based on http://www.theserverexpert.com/elastix_2.4_ha_cluster-updated.pdf |
08:43.51 | vNistelroot | ohh, thank you zpotoloom, let me check it ! |
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08:59.52 | vNistelroot | zpotoloom: took you a considered timeto set-up ? |
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09:00.12 | vNistelroot | or is straight-forward |
09:02.52 | vNistelroot | for a little feedback |
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09:55.30 | Stefan27 | does make samples only add files in /etc/asterisk ? |
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10:27.15 | vNistelroot | some feedback enabling srtp ? |
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11:55.46 | ruben23 | hi guys any help and suggestion....which is better setup... 1. office site ----> local asteriks gateway -------peer ---->asterisk/US (voice/data separation) 2. office site -------> remote asterisk same country------->peer ------->US asterisk -- which will genarte best result with audio quality guys ..please any suggestion |
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12:53.46 | sekil | hello |
12:57.25 | file | hi |
12:59.15 | sekil | I have a situation where my * 11.12.1 sends 180 ringing to b-leg uppon receiving 183 w/o sdp |
12:59.30 | sekil | makes my b-leg fake rings |
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13:07.33 | kb3ien | I'm getting some errors with func_odbc: I'm trying to debug backward from Unable to execute query [SELECT location FROM presence WHERE id='s'] |
13:15.41 | kb3ien | cli reports that `odbc read ODBC_PRESENCE s exec ` Failed to execute query. [SELECT location FROM presence WHERE id='s'] |
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13:23.04 | kb3ien | is there a way to make odbc more verbose ? |
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13:33.17 | jwr__ | i'm thinking about moving an on-site pbx to amazon. which means the connection between people's desk phones and the pbx would go over the internet. what are the best practices for that? SIP encryption? |
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13:39.13 | kb3ien | isql can connect as the asterisk user, but not asterisk.... grrr. |
13:42.18 | kb3ien | SIP and RTP encryption are good, but with some codecs or with VAD, you want to be careful. The size of the packets are not constant. and the information leaks that way. |
13:42.57 | kb3ien | you probably want to consider running IPSEC in addition to other precautions. |
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13:44.49 | vNistelroot | Hi all |
13:45.12 | vNistelroot | zpotoloom: I have a question about drbd, could you please point me in the right way? |
13:47.21 | vNistelroot | For my servers, I have a 160 GB of SAS hd for SO and 1TB SATA for recordings and so on |
13:48.02 | vNistelroot | as far as I´ve read, when partitioning you must create two partitions for the SO, and let one unmounted for drbd |
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13:49.36 | jepperl | Hey, im currently debugging an issue where the asterisk is sending RTP packets to another port than it announces to the client. This causes the server to sometimes not respect the range set in rtp.conf. There is no NAT (or PAT) involved in the setup on the server side |
13:50.26 | jepperl | Cant seem to find anything about it on the web, any chance some of you guys might have seen it before? |
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13:50.46 | cpugenius | does anybody know of a way to detect that a recall (disconnect) has occurred in the atxfer feature? |
13:50.55 | Penguin | I'm fairly sure the end point device is responsible for telling asterisk which RTP ports it wants to use. |
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13:51.32 | jepperl | In that case, asterisk does not comply with the client request |
13:51.35 | zpotoloom | vNistelroot: i'd recommend that you should really get familiar with drbd in general at first ( http://www.drbd.org/ ) before going near to any live systems |
13:52.08 | Penguin | What range are you setting on the phones, and what range are you setting on asterisk's rtp.conf? |
13:52.08 | wdoekes | jepperl: nat=auto/yes causes the client-set port to get ignored if the client sends from a different port first |
13:52.23 | zpotoloom | i'm not at home with drbd yet, thus can't give any good advice |
13:52.39 | wdoekes | symmetric rtp |
13:53.01 | zpotoloom | but the setup logic seems promising, and using the howto it worked |
13:53.06 | kb3ien | odbc seems to have no debugging at any level. |
13:53.16 | zpotoloom | just need to dig deeper |
13:53.26 | wdoekes | kb3ien: core set debug 8 |
13:53.30 | jepperl | penguin: i have tried with port > 1024 and 6000:30000 . In the rtp.conf i have 13000:65003 |
13:53.48 | wdoekes | and otherwise add tracing/debugging to unixodbc itself |
13:54.11 | jepperl | wdoekes: my serverside nat setting is force_rport;comedia |
13:54.24 | Penguin | That's the same as 'yes'. |
13:54.33 | kb3ien | yep it goinng to be a problem for another day.... |
13:54.44 | Penguin | If no nat is involved, set nat to 'no'. |
13:54.56 | wdoekes | jepperl: comedia == symmetric rtp |
13:57.45 | vNistelroot | thanks zpotoloom ! Im reading it deeply, im unsure how to manage partitions |
13:57.59 | vNistelroot | not the process but thinking about size |
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14:09.48 | jepperl | Thanks for the answers :) Is there any place on the webz where i can find information on all the dtls settings? (dtlssetup mostly).. Can't seem to find anything using google besides stackoverflow links with no real explanation |
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14:12.48 | superrafal | hi guys |
14:14.04 | superrafal | can anyone help me in getting WebRtc to work with asterisk 11.12? I'm using SipML5 library and I can't take it anymore. Too much problems ;) |
14:14.27 | jepperl | superrafal: what problems do you have? |
14:15.21 | superrafal | <PROTECTED> |
14:15.33 | superrafal | on menuselect in asterisk i don't see a module for that however |
14:15.43 | newtonr | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
14:15.54 | file | there is no module, it's automatically compiled in if your OpenSSL has DTLS-SRTP support |
14:16.07 | file | the configure script does a check for the support |
14:16.50 | superrafal | great idea, gonna check it out thanks :) |
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14:17.05 | newtonr | superrafal, see the dependencies in that tutorial |
14:18.27 | newtonr | jepperl, what about the documentation in the sample sip.conf ? |
14:18.37 | newtonr | jepperl, regarding your comment on dtlssetup |
14:19.25 | newtonr | jepperl, http://hastebin.com/luhuzopoka.vhdl |
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14:21.52 | jepperl | newtonr: found it :) thanks a ton |
14:22.37 | newtonr | jepperl, if there is ever documentation seemingly missing from either the sample config files or wiki.asterisk.org (the two official places) then please let me know so I can create an issue to go write it. |
14:22.49 | newtonr | jepperl, np |
14:24.19 | jepperl | newtonr: i certainly will do that :) neat |
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15:29.46 | babak | rmudgett: Hi, it it possible to use AOC in ECT service ? |
15:30.12 | babak | for tranfered calls |
15:30.44 | drmessano | With this recent BASH bug, I would like to move all my PBX'es to the more secure Windows platform. Who can help me install Asterisk in Windows? |
15:33.02 | Qwell | drmessano: Please to be trying http://www.asteriskwin32.com/ |
15:33.47 | drmessano | Most thank you |
15:34.17 | drmessano | You require support me?? |
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15:45.29 | WIMPy | babak: Not from Asterisk. |
15:48.17 | WIMPy | Actually Apart from passing it on, there's nothing you can do with AOC AFAIK. |
15:52.26 | babak | WIMPy: I mean on normal behavior Local exchange will send AOC for calls that Asterisk transfered completely to it with ECT service ? |
15:53.38 | WIMPy | It can do, yes. |
15:57.08 | gusto | what is AOC? that monitor company? |
15:57.35 | [TK]D-Fender | ~aoc |
15:57.51 | Qwell | gusto: Advice of Charge |
15:57.51 | [TK]D-Fender | Advice Of Charge : telcom billing signalling |
15:57.57 | gusto | ok |
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16:01.19 | WIMPy | imagines your telco sending you a TV or monitor each time you transfer a call. LOL |
16:01.30 | gusto | no |
16:07.15 | pryshebliad | is someone installed from this source https://build.opensuse.org/package/binaries/network:telephony:asterisk-12/asterisk?repository=Debian_7 ? |
16:08.03 | pryshebliad | Hey |
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16:30.11 | Stefan27 | is http://packages.asterisk.org/centos/6/asterisk-12/i386/RPMS/ likely to be updated with an rpm for 12.6.0 at some time in the future? |
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17:28.37 | babak | Hi, I want to use Asterisk just for playing costume music on hold(crbt), may be have more than 1000 different users , with musiconhold.conf 1000 classes is very hard ,how we can define dynamic music file names ? |
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17:43.49 | babak | or is it possible to pass a parameter form dialplan to music on hold class to choose specific file to play ? |
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18:49.12 | cpugenius | babak: you can use the realtime backend for musiconhold to have it backed by a database table instead |
18:49.31 | cpugenius | maybe create a view based on a users table with or without defaults |
18:52.54 | babak | cpugenius: thx, is it possible to simply put music file path in Dial(...m(/path/music.wav)) ? |
18:59.02 | rmudgett | babak: no The system works with musicclass names because a musicclass could be a media stream. |
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19:02.51 | babak | or maybe it was easyer to use simultaneous call with Dial(destination&musicExtention) |
19:03.40 | babak | and play music without Amswer on MusicExtention |
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19:11.05 | babak | <PROTECTED> |
19:11.23 | rmudgett | I don't know |
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19:15.49 | babak | rmudgett : thank you very much for your helps, I test successfully ECT , and now I am thinking to use Asterisk+ECT for providing CRBT for LocalExchanges subscribers |
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19:38.54 | zamba | where can i find the script for the different voice prompts for asterisk? i want to create a new language for it |
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19:41.47 | AkkerKid | Hi All! |
19:42.51 | zamba | hi you! |
19:43.16 | AkkerKid | is there a vicidial channel or does everyone hang out here? |
19:47.36 | AkkerKid | I'll take that as a "Meh" |
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20:23.16 | marceloamorim | should I keep safe_asterisk running if I receive the message unable to connect to remote asterisk ( does /var/run/asterisk/asterisk.ctl exist? I remove all modules run asterisk with -vvvc and didn`t get any error |
20:23.59 | marceloamorim | the permission to that pid file is the same to the /etc/asterisk |
20:24.26 | marceloamorim | and appears just if I run asterisk -c or safe_asterisk |
20:40.09 | newtonr | sounds like you might have another funky asterisk process running at the same time |
20:41.30 | marceloamorim | well, I uninstall the version 11 and install this ast 12.6 to test the cel |
20:42.04 | newtonr | so what does "ps aux | grep -i asterisk" show? |
20:42.26 | marceloamorim | nothing |
20:42.32 | marceloamorim | just the grep |
20:43.21 | newtonr | yeah then probably still a permission issue. triple check everything! |
20:45.56 | marceloamorim | I didn`t find any asterisk.service to run with this systemctl |
20:46.22 | marceloamorim | but the [ OK ] appears, but I`ll try, ty newtonr |
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21:14.34 | Katty | drmessano: YOU |
21:14.44 | Katty | drmessano: txt me. |
21:14.50 | Katty | drmessano: cause i'm afk now. |
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21:19.14 | WIMPy | feels reminded of late night advertising. |
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22:28.37 | ovoshlook | Hello, does anyone know about default call limits at asterisk? I try to do load test between 2 asterisks via originate, but when I send 100 calls, I see maximum 38 channels at asterisk that recieved calls, another calls queued as I think. Am I right with at my guess? |
22:29.35 | WIMPy | There's no limit. |
22:29.49 | WIMPy | But chan_sip is good at blocking itself. |
22:29.49 | Skten | Unless there is some limit / issue in originate, there are no out of the box "limits" |
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22:47.19 | fornax | Hi I'm using asterisk 12.5 and need a hint how I can listen for a devstate change via ari websocket, can anyone help me? |
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23:01.54 | wrxed | trying to enable dnsmgr in asterisk 11. I have set the enable=yes in dnsmgr.conf yet when I do dnsmgr status it says disabled... Am I missing something obvious (i've restarted asterisk since modifying the file as well as reloaded the module) |
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