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| 05:53.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:18.11 | fcalderero2 | Hello, can you help me? i have a tls sip trunk with a provider, the certificate options are set in sip.conf and now i need to connect to a second provider whit different tls options. Can i use an additional tls certificate for this second tls trunk? Thanks a lot |
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| 10:33.16 | GreatSUN | hi all |
| 10:34.08 | GreatSUN | I have moved to a new house, got a fritzbox 7490 with voip connection and need to setup my asterisk to use it properly |
| 10:34.30 | GreatSUN | also I have been setting up a new sip client on my windows computer microsip |
| 10:35.22 | GreatSUN | now after I think everything is properly configured, I can call outside without tone and calls from outside are not delivered properly to asterisk |
| 10:35.39 | GreatSUN | sip show registry says properly registered |
| 10:35.59 | GreatSUN | sip show peers says status OK |
| 10:36.23 | GreatSUN | any idea what might be wrong in my configuration? |
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| 13:49.22 | fcalderero2 | thanks |
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| 15:03.22 | RadJAckson | Hi , my question isnt related to Asterisk specifically , i need to know what solution is there on the market to leave a message directly on the answer machine ? there are a couple of services here in france doing it , i think they have a direct link with major telephony providers to have access to such service , isnt there any workaround ? |
| 15:04.57 | [TK]D-Fender | "the answer machine" is VERY bad use of terminology |
| 15:05.04 | [TK]D-Fender | VM isn't a singular service |
| 15:05.06 | [TK]D-Fender | WHOSE Vm? |
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| 15:06.40 | RadJAckson | Well ive got a coupe of links , campanies here in france offering short numbers to call (4 digits), once the call is answered you type a phone number , and then you access directly to the answer machine |
| 15:06.42 | RadJAckson | and its legal |
| 15:08.12 | [TK]D-Fender | Then you already have the means |
| 15:08.19 | [TK]D-Fender | you dial the access number and the box |
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| 15:10.16 | Nightwolf | hi |
| 15:11.13 | RadJAckson | do you think that they pay to be granted access to the customers voice mails ? like a subscription to the telephone network operator ? |
| 15:12.10 | [TK]D-Fender | If it isn't owned by them then you can be sure there is money changing hands |
| 15:14.30 | GreatSUN | Hey all |
| 15:14.56 | GreatSUN | I still have problems connecting asterisk properly to my fritzbox 7490 |
| 15:15.13 | GreatSUN | outgoing call works properly (without any sound/tone) |
| 15:15.20 | GreatSUN | incoming doesn't work at all |
| 15:15.33 | GreatSUN | any ideas what is wrong? |
| 15:16.31 | Nightwolf | i'm new to asterisks and administrate a system i didn't install. i get the following error when dialing a specific number (works on other numbers): |
| 15:16.47 | Nightwolf | Got SIP response 480 "Temporarily Not Available" back from 10.10.10.161 |
| 15:17.03 | Nightwolf | <PROTECTED> |
| 15:17.15 | Nightwolf | == Everyone is busy/congested at this time (1:0/1/0) |
| 15:17.18 | Nightwolf | any idea? |
| 15:17.54 | Rico | is there a reason why when asterisk sends a SIP INVITE to an UAC, this INVITE does not have a "Route" header ? |
| 15:18.20 | Rico | asterisk 13 / pjsip |
| 15:18.21 | RadJAckson | [TK]D-Fender I've tried one alternative before and it worked :) I create two auto dial call files , calling the same number , i copy the first one to the outgoing directory , 1 second later i copy the second. it worked , the first call file place the line in busy mode the second one falls directly into the answering machine , it worked with 4 major telephony operators , the 5th one blocked |
| 15:18.22 | RadJAckson | it :) |
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| 15:23.27 | WIMPy | GreatSUN: A little more information could help. But why do you use Asterisk with a FB and not the other way round?> |
| 15:27.24 | GreatSUN | cause provider has automated configuration on fritzbox which I don't know |
| 15:27.37 | GreatSUN | and where I don't know how I could adapt this at my asterisk |
| 15:27.44 | GreatSUN | the FB is also the internet router |
| 15:27.49 | WIMPy | Download the config. |
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| 15:35.00 | GreatSUN | WIMPy: got it, a lot of stuff in there for the voip configuration |
| 15:35.13 | GreatSUN | and I dunno how to adapt this to my asterisk now :-( |
| 15:36.22 | WIMPy | Plenty of information on Google. And here as well if there are questions left. |
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| 16:04.22 | Rico | I'm still trying to understand why asterisk re-negociates to g711 channel after T38 fax completion (just to close the call ?) |
| 16:04.25 | Rico | any idea ? |
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| 16:25.42 | wdoekes | Rico: there may be no other reason than "the reference implementation did that" |
| 16:26.15 | Rico | wdoekes: so my question should be : "why do asterisk does that ?" ;) |
| 16:27.10 | Rico | did not see that in RFCs |
| 16:27.48 | Rico | (https://www.ietf.org/proceedings/55/I-D/draft-ietf-sipping-realtimefax-00.txt) |
| 16:28.12 | Rico | mmmh |
| 16:28.16 | Rico | page 14/15 |
| 16:29.11 | wdoekes | "4. Once the fax transmission is terminated, audio capabilities are �restored� or the call is terminated. |
| 16:29.15 | wdoekes | (or both :P ) |
| 16:29.23 | Rico | <PROTECTED> |
| 16:29.23 | Rico | <PROTECTED> |
| 16:29.23 | Rico | <PROTECTED> |
| 16:29.24 | Rico | yes |
| 16:29.26 | Rico | right |
| 16:29.30 | Rico | did not see that |
| 16:29.31 | Rico | thanks |
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| 16:35.20 | Tagor | is it possible to use a different outgoing ip address for a peer? |
| 16:37.03 | [TK]D-Fender | Tagor: * uses whatever your routing table says |
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| 17:39.36 | zekoZeko | anyone seen this: ATA (used atcom and Cisco SPA112) rings the phone once, then picks up the line for a second (silence) and hangs up. |
| 17:40.03 | zekoZeko | this happens when analog phones have batteries (for called id display etc). If I remove the batteries, everything works as should. |
| 17:40.10 | zekoZeko | called=caller |
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| 18:58.40 | spicyramen_ | hi * people, How to change the s= line in SDP, currently I get: s=Asterisk PBX UNKNOWN__and_probably_unsupported |
| 18:58.43 | spicyramen_ | and need to hide it |
| 19:04.48 | file | if using chan_sip that is configured using the sdpsession configuration option |
| 19:23.29 | *** join/#asterisk nny (4bb09776@gateway/web/cgi-irc/kiwiirc.com/ip.75.176.151.118) |
| 19:24.54 | nny | man are you guys gonna love this question. Long story short working on building 1.2 and zaptel in CentOS7, there is a request to upgrade/update to 11 but right now custom modules and apps need to be modified, removed or elsewise. All is good with asterisk 1.2 and whatnot, it's running, no big deal. Zaptel is fighting me though. |
| 19:25.39 | nny | I had to symlink some kernel folders (i.e. asm) to get it to get to the point where it breaks fantaastically during compile, testing both their old zaptel source and the latest 1.4.12. |
| 19:26.14 | nny | at this point it looks like compiler syntax, endiness etc breaking things |
| 19:26.45 | nny | good news is all that I think is needed is dummy for timing purposes |
| 19:27.10 | nny | so.. just poking around at suggestions. This is a stop gap until update |
| 19:33.51 | nny | making some progress, currently testing with 1.4.12, working on something like this issue. http://forums.digium.com/viewtopic.php?t=71235 |
| 19:38.47 | nny | my questioni s actually more along the lines of "ztdummy for 1.2, is it just for conference?"... i am beginning to wonder if they really need it here. As far as my knowledge goes it's only used for conf, but IIRC it may be used for some bridging too |
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| 19:43.26 | malcolmd | app_meetme requires dahdi (zaptel), along with anything that uses app_meetme, like old versions of app_page, or SLAstation / SLAtrunk; or other things that require timing, such IAX2 w/ trunking |
| 19:43.34 | *** join/#asterisk jordanfeld (~jordan@66.23.136.139) |
| 19:43.37 | malcolmd | that's so far in the past though, that it's not still clear |
| 19:44.48 | jordanfeld | Hello All, for those that have worked with AGI..I am running into an issue setting a var. |
| 19:44.58 | jordanfeld | I am trying to do "SET VARIABLE "CONFBRIDGE(user, music_on_hold_class)" "inherit"" |
| 19:48.03 | jordanfeld | I am able to set other vars |
| 19:49.28 | robmal | I'm pretty sure its getting angry about that between 'user,' and 'mohclass' |
| 19:50.28 | jordanfeld | That is the way I have it set in the dial plan. Would it be wise to add quotes for AGI ? |
| 19:50.45 | nny | malcolmd: Thanks yeah, I had to mod zaptel-base.o to use unlocked_ioctl for zt_ioctl and i am down to one error, however the other issue with proc_fs.h will require more work. This is just getting zaptelbase to compile, the compiler hasn't hit ztdummy yet (i disabled everything else in make menuconfig to test). |
| 19:51.33 | nny | current status. http://pastebin.com/Mk03WwQa in the end I am going to pause and check need for zaptel in this setup |
| 19:52.03 | nny | surprised how easily asterisk 1.2 compiled although we have yet to stress test it |
| 19:52.14 | nny | then again it's not installing drivers like zaptel does |
| 19:52.20 | nny | modules/drivers |
| 19:54.13 | nny | anyways thanks malcolmd. I will go that route first, if I can avoid zaptel use until updating to them to 11 I will |
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| 19:56.53 | malcolmd | good luck :) |
| 19:57.56 | nny | haha :D |
| 20:01.04 | malcolmd | iknowright? ;D |
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| 20:01.48 | gruetzkopf | oh f*, the MO drive on my PBX just broke |
| 20:03.25 | *** join/#asterisk spicyramen_ (~Adium@173.227.7.2) |
| 20:03.42 | spicyramen_ | hi * people, does asterisk 14.3 support RTP to SRTP ? |
| 20:03.46 | spicyramen_ | 13.4* |
| 20:04.03 | mjordan | all versions that support SRTP (which is 1.8+) would support that. |
| 20:04.05 | spicyramen_ | sip phone rtp â> * â> TLS SIP provider |
| 20:04.06 | mjordan | Asterisk is a B2BUA |
| 20:04.29 | spicyramen_ | this TLS SIp provider do SRTP/TLS |
| 20:04.35 | spicyramen_ | my phone only does RTP |
| 20:04.37 | spicyramen_ | TCP |
| 20:05.35 | spicyramen_ | I get Encrypted signalling is required in cha_sip/c6182 |
| 20:05.45 | spicyramen_ | chan_sip:6182 |
| 20:05.51 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 20:07.18 | spicyramen_ | TCP(RTP) â * -> TLS (SRTP) is thi supported @mjordan |
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| 20:29.26 | mjordan | yes |
| 20:32.03 | JoshaX | Is it possible to run multiple process (load balancing 3 servers) that connect to a single Asterisk server using AMI connection? |
| 20:46.22 | WIMPy | What processes? |
| 20:46.27 | WIMPy | Your question is more than vague. |
| 20:49.21 | Ice_Strike | A process that run in the background on linux. Each process connect to AMI |
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| 20:50.42 | robmal | AFAIK there's no limit to ami connections, so yes. |
| 20:51.10 | WIMPy | Indeed |
| 20:51.59 | Ice_Strike | robmal So Let say I originate a call from "Process A" - will the reponse show the same time for all Processes? |
| 20:52.27 | WIMPy | time? |
| 20:52.44 | WIMPy | Only the one sending a command will get a responese. |
| 20:52.56 | WIMPy | But they will all get te events that follow. |
| 20:53.16 | robmal | AFAIR it only responds to the user sending the request, but you can get some clues from events that show up on all the connections. |
| 20:53.55 | robmal | [Note to self: shorter sentences ] ;-) |
| 20:55.43 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:58.16 | Ice_Strike | Yea I will need to test it. |
| 20:59.06 | Ice_Strike | Have you ever done Node.JS load balancing for AMI processes? |
| 20:59.34 | WIMPy | I'm not sure what that's supposed to do. |
| 20:59.50 | robmal | Lets skip the node.js part and tell us what do you want to achieve. |
| 21:00.03 | WIMPy | Or why you'd need to load-balance anything that connects via AMI. |
| 21:00.25 | WIMPy | Well, at least it's not Java... |
| 21:01.28 | robmal | In .pl there was once a job offer which required 'Java (especially the Script version)' ;-) |
| 21:01.49 | robmal | I still don't know if it was fake. |
| 21:02.58 | WIMPy | Lets not start senseless job offers. |
| 21:05.29 | robmal | Ice_Strike: So? |
| 21:05.58 | Ice_Strike | WIMPy I am just exploring some architecture ideas in regarding scaling AMI application if needed in the furture. Let say there is 5 Asterisks box and I want a single process (or load balancing) to communicate all those asterisk servers via AMI. |
| 21:07.45 | robmal | I could imagine 5 boxes shooting AMI requests at one * server, but why should 5 * boxes do the same thing at the same time? Just pick one and check if it responds success. |
| 21:10.42 | Ice_Strike | robmal 5 asterisk boxes is for load balancing if agents grow |
| 21:11.39 | WIMPy | Ok, why not. |
| 21:11.40 | robmal | Ok, so go round-robin, check channel usage or whatever and adjust the weight accordingly. |
| 21:12.25 | WIMPy | You couls also coleect system load data via other channels. |
| 21:13.11 | Ice_Strike | and then most complicated part is shitty asterisk queue |
| 21:13.15 | WIMPy | once did load balancing of web servers based on system load. |
| 21:13.42 | WIMPy | If you don't like it, don't use it. |
| 21:15.34 | robmal | If you want to go full hardcore - Make a queue on one * box and on pickup channelredirect() caller and agent to one of the other 4 boxes. |
| 21:16.02 | robmal | I've never tried it, just because nobody asked hard enough, but i think it should work like a charm. |
| 21:17.28 | Ice_Strike | I have read about Proxy Server (register agents/queues on multitple asterisk servers) |
| 21:18.26 | Ice_Strike | Like kamailio |
| 21:18.52 | jordanfeld | Hey guys. Sorry for reposting this but I was knocked offline earlier. |
| 21:19.18 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 21:19.21 | jordanfeld | I am unable to issue the AGI command SET VARIABLE "CONFBRIDGE(user, music_on_hold_when_empty)" "yes" |
| 21:19.53 | jordanfeld | Someone pointed out it might have something to do with (user, music_on_when_empty) I am not sure what they meant |
| 21:21.06 | Ice_Strike | robmal So let say I have 5 asterisks for load balancing when agents grow. It is not harm having load balancing or 2 process of AMI process to connect with these asterisk boxes? |
| 21:26.46 | robmal | http://leifmadsen.com/sites/default/files/cc01-madsen-digium.pdf |
| 21:27.03 | robmal | This is a good lecture on the subject. |
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| 21:31.49 | jordanfeld | Does anyone know of a good example of someone using AGI to setup a confbridge instead of flat files ? |
| 21:32.04 | jordanfeld | Are their any obvious limitations ? |
| 21:32.21 | Ice_Strike | robmal i will check it out |
| 21:33.21 | robmal | jordanfeld: There are non, i'm using AMI to dynamically create and start scheduled conferences. Try harder, confbridge is more fun than meetme. |
| 21:34.12 | robmal | Ice_Strike: If you try my idea of ChannelRedirectEverything() please let me know how dumb it is in real life ;-) |
| 21:36.19 | *** join/#asterisk jordanfeld_ (4217888b@gateway/web/freenode/ip.66.23.136.139) |
| 21:36.36 | jordanfeld_ | I am using confbridge. And I am trying to repalce the dialplan with FastAGI |
| 21:36.42 | Ice_Strike | robmal Lets see |
| 21:37.10 | Ice_Strike | robmal Just reading that pdf, it say "Can login to queues across the cluster" |
| 21:37.17 | Ice_Strike | How does that work? |
| 21:38.51 | [TK]D-Fender | jordanfeld_, show us the actual call & failure |
| 21:39.35 | [TK]D-Fender | jordanfeld_, And AGI has nothing to do with setting up ConfBridge. DIALPLAN APPS do. And it works the same inside as outside AGI. |
| 21:39.53 | [TK]D-Fender | jordanfeld_, The apps & functions do the same job. Question being : are you actually doing it right? |
| 21:40.03 | [TK]D-Fender | jordanfeld_, So show us your real call debug for this |
| 21:40.15 | jordanfeld_ | I dont exactly have a failure. I am tring to set CONFBRIDGE vars before I issue a EXEC CONFBRIDGE |
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| 21:41.37 | [TK]D-Fender | Show us the actual call. |
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| 22:02.19 | jordanfeld_ | Here is the actual call http://pastie.org/10342698 |
| 22:02.49 | jordanfeld_ | An earlier log show "[2015-08-10 21:21:18] WARNING[17797][C-0000011d] confbridge/conf_config_parser.c: CONFBRIDGE(user, dsp_drop_silence) cannot be set to 'yes'. Invalid type, option, or value." |
| 22:05.24 | [TK]D-Fender | you have not enabled AGI debug |
| 22:05.29 | [TK]D-Fender | that output shows us nothing |
| 22:05.36 | [TK]D-Fender | "AGI <tab>" |
| 22:13.59 | [TK]D-Fender | is out of time for now and heads out for a bit. |
| 22:14.04 | jordanfeld_ | http://pastie.org/10342712 |
| 22:15.12 | [TK]D-Fender | I see no error in there |
| 22:15.41 | [TK]D-Fender | I see a LOT of extra quotes |
| 22:15.46 | [TK]D-Fender | Which is NOT good...... |
| 22:15.52 | [TK]D-Fender | And useless whitespace |
| 22:16.19 | jordanfeld_ | So when I start a confbridge why doesnt it pickup those vars? |
| 22:16.37 | [TK]D-Fender | My guess = all those extra garbage quotes |
| 22:17.00 | [TK]D-Fender | "'CONFBRIDGE(user, music_on_hold_when_empty)'" "yes" |
| 22:17.01 | robmal | jordanfeld_: Please try SET VARIABLE CONFBRIDGE(user,wait_marked) YES |
| 22:17.05 | [TK]D-Fender | single + double everywhere |
| 22:17.11 | [TK]D-Fender | BAD |
| 22:17.21 | robmal | No whitespace, no quotes, just SET VARIABLE VARIABLENAME VARIABLEVALUE |
| 22:17.38 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AGICommand_set+variable |
| 22:17.43 | [TK]D-Fender | Nowhere in there do you see quotes |
| 22:18.02 | [TK]D-Fender | actually heds out |
| 22:18.18 | jordanfeld_ | If VARIABLENAME is CONFBRIDGE(user, music_on_hold_class) wont that space mess it up ? |
| 22:18.27 | jordanfeld_ | or should I just remove that space |
| 22:18.50 | robmal | Jut remove the spaces in variable names and quotes around everything. |
| 22:18.50 | WIMPy | remove it |
| 22:19.01 | jordanfeld_ | Okay will do. |
| 22:24.36 | [TK]D-Fender | No undue whitespace, no extra quotes |
| 22:25.02 | [TK]D-Fender | and my area just got hit with a sun-shower. looked clear, got flushed. No biking tonight it seems |
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| 22:35.01 | jordanfeld_ | So it was the space between user, music_on_hold_when_empty that was killing it |
| 22:35.35 | WIMPy | Spaces are generelly dangerous. |
| 22:35.51 | jordanfeld_ | Lesson learned :) |
| 22:36.11 | jordanfeld_ | Thank you [TK]D-Fender and WIMPy |
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| 23:59.00 | zekoZeko | any recommendations on a sbc with gpio to run some automation with asterisk? Something like raspberry pi feature wise, but more reliable. No need for video, serial is fine for console. |
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