| 00:12.03 | *** join/#asterisk u0m3 (~u0m3@188.27.74.212) |
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| 00:57.14 | denysonique_ | hi |
| 00:57.34 | denysonique_ | Default Asterisk installation segfaults on Ubuntu 16.04. Is this a known bug? |
| 00:57.52 | denysonique_ | asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. |
| 00:57.52 | denysonique_ | Aborted (core dumped) |
| 01:03.59 | *** part/#asterisk catphish (~J@unaffiliated/catphish) |
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| 01:50.43 | Akari | What version? |
| 01:51.05 | Akari | And how are you installing it... |
| 01:51.37 | *** join/#asterisk bmg505 (~leon@196-210-200-224.dynamic.isadsl.co.za) |
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| 04:04.59 | denysonique_ | Akari, I solved the problem by disabling a module |
| 04:05.18 | denysonique_ | Akari, now I am testing it on another machine which did not require the module to be disabled |
| 04:05.37 | denysonique_ | Akari, but the problem I am having now is that a SIP provider is not getting registered... |
| 04:06.07 | denysonique_ | in show peers, I get status OK |
| 04:06.10 | denysonique_ | for the peer |
| 04:06.16 | denysonique_ | Asterisk 13 |
| 04:06.32 | denysonique_ | I am registering it with the register => ;statement |
| 04:06.44 | denysonique_ | wireshark does not report any SIP registratoin |
| 04:06.55 | denysonique_ | only an OPTIONS request |
| 04:07.05 | denysonique_ | which I guess is being sent by 'qualify=yes' |
| 04:07.57 | Samot | Can you confirm on your provider's side that the trunk is registered? |
| 04:09.42 | denysonique_ | On the providers side I do not see any asterisk user-agent being registered |
| 04:10.18 | denysonique_ | i.e. the web interface of the VOIP provider |
| 04:11.59 | Samot | OK, so open a sip debug on Asterisk and force it to register again. |
| 04:12.48 | denysonique_ | sip show registry -- 0 SIP registrations. |
| 04:12.48 | denysonique_ | hmm |
| 04:12.53 | denysonique_ | Samot, how do I force it to register |
| 04:12.56 | denysonique_ | sip reload? |
| 04:13.01 | Samot | Yes. |
| 04:13.27 | *** join/#asterisk rpifan (~rpi@73.106.73.75) |
| 04:14.32 | denysonique_ | Samot, it only reports the OPTIONS sip protocol request, the same thing I can see in wireshark |
| 04:14.36 | denysonique_ | with a 200 request from the provider |
| 04:14.44 | denysonique_ | asterisk does not send a REGISTER request |
| 04:15.10 | denysonique_ | this is the first time I am configuring asterisk |
| 04:15.10 | Samot | Show your peer details, mask any passwords/secrets. |
| 04:15.13 | Samot | ~pb |
| 04:15.13 | infobot | methinks 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:15.16 | Samot | ^ Show it there. |
| 04:19.38 | denysonique_ | Samot, http://hastebin.com/ayewimabop.http |
| 04:20.42 | Samot | OK, so you don't need to allow= fields. |
| 04:21.01 | Samot | And I highly doubt that your provider is behind NAT. So nat=yes is not correct. |
| 04:22.57 | denysonique_ | I thought nat=yes related to me |
| 04:23.20 | denysonique_ | however after reloading without the nat option the providers is UNREACHABLE, as reported by show peers |
| 04:23.39 | denysonique_ | is sip.conf the only file I need to edit in order for asterisk to perform a registration? |
| 04:29.24 | Samot | UNREACHABLE means Asterisk is not getting the replies from the other side. |
| 04:29.38 | Samot | Either because they never make it there or because it's being sent back to the wrong place. |
| 04:29.46 | Samot | What did the SIP debug show? |
| 04:33.27 | denysonique_ | Samot, only 'retransmitting' and then the sip packet being sent |
| 04:33.31 | denysonique_ | requesting the OPTIONS |
| 04:33.46 | denysonique_ | if I switch back nat I get a response for the OPTIONS request sent by asterisk |
| 04:34.07 | Samot | So retransmitting means it never got a response. |
| 04:34.12 | Samot | For whatever reason. |
| 04:34.20 | Samot | Is Asterisk behind NAT? |
| 04:36.22 | denysonique_ | yes it is behind NAT, however if I switch nat=yes |
| 04:36.32 | denysonique_ | it successfully connects and sends the OPTIONS sip request |
| 04:36.42 | denysonique_ | for which it gets a requests, and shows as online in 'show peers' |
| 04:36.49 | denysonique_ | 'sip show peers' |
| 04:37.46 | denysonique_ | even in wireshark I see no attempts of asterisk trying to register |
| 04:37.53 | denysonique_ | only sending the OPTIONS |
| 04:39.02 | Samot | Do you have your external IP and local_net settings? |
| 04:43.08 | denysonique_ | Samot, nope |
| 04:44.35 | Samot | Well that's a problem. |
| 04:44.53 | Samot | Those tell Asterisk that *it* is behind NAT. |
| 04:45.38 | Samot | Your sending your private IP in the contact details with nothing in the SIP message to show the received path. |
| 05:00.13 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 05:08.45 | masked | doesn't asterisk have any nat or upnp detection? |
| 05:09.00 | masked | cos it seems awefully flaky |
| 05:11.35 | Samot | Yes. |
| 05:11.54 | Samot | You can set nat= with the three or four NAT options in the general section of sip.conf |
| 05:12.21 | Samot | By "default" it assume NAT based on the RFC. |
| 05:12.36 | Samot | But if you don't tell it what the external IP details are, it won't know. |
| 05:12.55 | Samot | Then it needs to know what networks it should consider local. |
| 05:12.59 | Samot | So you need to tell it that. |
| 05:14.31 | masked | yeah i think the latter is more to the point the rest is trivial, it wont work otherwise.. |
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| 05:43.52 | masked | Samot: what if, regardless of nat, i want to go back through a proxy... not just sending registrations with outbound_proxy, but all responses? |
| 05:44.20 | *** join/#asterisk Ahmadi (02b14b1f@gateway/web/freenode/ip.2.177.75.31) |
| 05:44.27 | Ahmadi | Hi |
| 05:44.41 | Ahmadi | I have registration failed for one of my extension. |
| 05:44.58 | Ahmadi | i need details of the registration failed to debug |
| 05:45.11 | masked | hmm |
| 05:45.15 | masked | i might just wait till tomorrow |
| 05:45.22 | masked | i can register with my upstream with opensips |
| 05:45.39 | Ahmadi | for example user is wrong, password is wrong or ip is bane or ... |
| 05:45.41 | masked | and i can auth if i put this patch in the right place... |
| 05:45.53 | masked | then just rewrite to them and then i wont need my own asterisk |
| 05:46.19 | masked | Ahmadi: are you using pjsip or sip? |
| 05:46.24 | Ahmadi | pjsip |
| 05:46.52 | masked | run asterisk -r and core set verbose 4 and debug 4, and pjsip set logger on |
| 05:46.55 | *** join/#asterisk SaintPeteJackBoy (~IceChat9@70-126-192-156.res.bhn.net) |
| 05:47.16 | masked | then make a registration attempt and check see auth failed |
| 05:49.06 | Ahmadi | i did |
| 05:49.11 | Ahmadi | i have no log |
| 05:49.22 | masked | oh well |
| 05:49.32 | Ahmadi | seems that registration never reach to asterisk |
| 05:49.41 | Ahmadi | i mean registration request |
| 05:50.29 | masked | hah yeah |
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| 06:37.02 | WIMPy | That's a new one. Have teh telemarketers finally discovered SIP or why is someone trying to bute force call any 3 or 4 digit number? |
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| 06:47.25 | masked | im getting a 407 from my upstream via my proxy the from header says user@proxy to dest@upstream, is it failing because my user is trying to auth against my upstream? |
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| 06:59.49 | duper | how do I get the environment variables I see when I run `cmd.exe /c set` from other processes in the task list? |
| 06:59.59 | duper | err wrong channel |
| 07:00.46 | duper | I meant to ask.. are there still SIP termination services that allow you to craft outgoing call data as VXML? |
| 07:01.58 | duper | also, is there a SIP CLEC/broker/whatever that specializes in MoIP/FoIP (Modem over VoIP and Facsimile over VoIP) |
| 07:03.15 | duper | trying to hook up a Livingston Portmaster so ppl can POTS into old boxen I have.. |
| 07:03.34 | duper | it's ComOS which I don't think is even used anymore |
| 07:03.45 | WIMPy | The good old portmaster |
| 07:08.04 | duper | it's a 24-port :P |
| 07:08.12 | duper | serial ports |
| 07:08.31 | duper | with an ethernet for console |
| 07:08.37 | WIMPy | Oh the little one. |
| 07:08.50 | duper | i'm sniffing traffic from it and it still has the same IP address it was using in the late 90's |
| 07:09.13 | duper | yah, i suppose.. there were several of them in the ISP I used to work at |
| 07:09.30 | duper | maybe 4 iirc .. two were lucent and the other two livingston |
| 07:09.45 | duper | which went to radius on linux |
| 07:09.57 | duper | anyway |
| 07:10.31 | duper | where can i find a list of the pros and cons of various SIP termination (and/or origination) providers? I'm not sure who to go with.. |
| 07:12.51 | duper | WIMPy: an interesting thing about dialing into PPP on a portmaster like that is there's a packet it supports which is in CHAP/PAP format that translates into a DHCP request (so the person dialing in can get a certain IP if it's available, regardless of if static or dynamic IP's are in use) |
| 07:13.27 | duper | I dialed into SLIP before that, I just don't know what kind of hardware it was cause I never saw it.. |
| 07:13.52 | WIMPy | You can always request a certain IP via PPP. |
| 07:14.07 | duper | do you know what the dip command is? |
| 07:14.27 | duper | i haven't used it in so long.. i'd have to go thru the docs to find that for sure |
| 07:14.32 | WIMPy | dip? |
| 07:14.43 | duper | it's like a PPP daemon for Linux |
| 07:14.48 | duper | err |
| 07:14.54 | duper | well, it's a dialup scripter |
| 07:15.06 | duper | but does IPC with pppd or something.. not sure |
| 07:15.28 | WIMPy | don't know that script the ppp parameter is just localIP:remoteIP. |
| 07:16.02 | duper | http://www.pegasusinfocorp.com/web_development_knowledgebase/linuxman/man0324.htm |
| 07:16.11 | duper | parameter to what? |
| 07:16.18 | WIMPy | pppd |
| 07:16.28 | duper | oic |
| 07:16.33 | duper | but doesn't that translate into some protocol data? |
| 07:16.54 | WIMPy | Sure |
| 07:16.59 | duper | what is it |
| 07:17.30 | duper | special PPP or ICMP packet? |
| 07:17.39 | WIMPy | Some ipcp thing. |
| 07:17.43 | duper | althogh I guess that would happen before the IP stack was even able to get itself established actually |
| 07:17.56 | WIMPy | yes |
| 07:19.18 | denysonique_ | What could be the reason of Asterisk not sending a REGISTER packet request? |
| 07:19.22 | denysonique_ | a fresh install |
| 07:19.38 | denysonique_ | it only sends a OPTIONS request for which it receives a reply |
| 07:19.42 | denysonique_ | as inspected by wireshark |
| 07:20.04 | duper | ppl use point-to-point so loosely as a routing scenario, that it kind of loses it's meaning, so i forgot it was like a protocol stack of its own |
| 07:20.28 | duper | denysonique_: that actually sounds like Content Security Policy of some kind |
| 07:21.06 | duper | but you're asking about SIP as opposed to HTTP, rite? |
| 07:21.13 | denysonique_ | yes about SIP |
| 07:21.14 | duper | not sure if SIP has CSP and friends.. |
| 07:23.08 | duper | Is asterisk being packaged with any sort of WebRTC samples yet? |
| 07:23.16 | duper | or a project asterisk is related to? |
| 07:23.41 | denysonique_ | in FreePBX distro everything is just working. In a raw asterisk install not... |
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| 08:25.40 | [TK]D-Fender | denysonique_, Show us why it should send one... |
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| 12:49.51 | dan_j | Hi. When a queue member is dynamically added to a call queue, is that membership saved something to survive asterisk restarts? |
| 13:00.07 | kunwon1 | only if you have realtime queues |
| 13:00.30 | kunwon1 | if you don't have a queue_member table somewhere (could have a different name) or some other mechanism for saving them manually, they will not be saved |
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| 13:20.43 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:23.24 | dan_j | ok. thanks |
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| 16:51.59 | spearo_ | hi |
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| 19:02.45 | spearo_ | :P |
| 19:03.23 | denysonique_ | Is it possible to invite someone into a confbridge conference from a conference dialplan? |
| 19:03.43 | denysonique_ | e.g. I press 1 and asterisk makes a call adding someone to the conference |
| 19:08.23 | masked | chop chop |
| 19:13.04 | [TK]D-Fender | Do you se any option in the conference that allows you to execute anything at all? |
| 19:17.37 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration |
| 19:17.41 | [TK]D-Fender | VERY fine manuals |
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| 19:22.22 | [TK]D-Fender | NEXT!@!!@ (c) BKW |
| 19:22.41 | spearo_ | Any idea why on some versions of 13.x.x, the Record() application does not delete the recording after a single hangup, yet on other versions it does? I'm aware that that as of (I believe 11) option k or y is suppose to be the definitive. However - for my applications of Asterisk - I find this mildly frustrating. |
| 19:23.19 | [TK]D-Fender | Delete recording? |
| 19:23.28 | [TK]D-Fender | Why would it delete the recording after the call ends exactly? |
| 19:27.31 | spearo_ | Not a call per say - just dialling an extension using the Record application. From what I can tell in asterisk 13.x.x, depending on what explicit version, if using the record application and hanging up - your recording will be there. On other versions, it will delete the recording unless you use the appropriate option for the application. im my scenerios - with newer current veresions of Asterisk - We cannot stop the recordi |
| 19:27.38 | spearo_ | mind the spelling |
| 19:28.05 | masked | who gives a flying fuck |
| 19:28.12 | masked | they can't use it as evidence in court |
| 19:28.35 | [TK]D-Fender | .... |
| 19:28.36 | masked | sicko's |
| 19:28.38 | spearo_ | thanks for the info 'masked' |
| 19:28.44 | masked | sorry |
| 19:28.46 | [TK]D-Fender | that makes no actual sense |
| 19:28.49 | masked | bit hyper this morning |
| 19:29.15 | spearo_ | in lehmas terms: dial extension -> make recording |
| 19:29.18 | [TK]D-Fender | Record doesn't record a conversation between 2 parties. |
| 19:29.42 | spearo_ | there is no recording of two converstaions |
| 19:30.24 | [TK]D-Fender | it's a dumb recording like when you are leaving voicemail, and is user-terminated |
| 19:30.34 | spearo_ | ye |
| 19:30.43 | [TK]D-Fender | Anyway I do recall there being options for what to do on hangup... do they not work as advertised? |
| 19:31.01 | spearo_ | they do - as advertised - for some of the later versions of 13.x |
| 19:31.11 | spearo_ | but not for the earlier 13.x. versions |
| 19:31.59 | [TK]D-Fender | doesn't really make sense... there's virtually no reason for the code of that app to even change. It's not like there's real functioanlity to add to what this was meant for.. |
| 19:32.12 | spearo_ | exactly what i thought |
| 19:33.10 | spearo_ | I've come across one project where this application is exactly what I require - but works differently on the same versions |
| 19:33.22 | spearo_ | I'm not looking for a resolve, only the need to vent the mild frustration :) |
| 19:40.26 | [TK]D-Fender | Oh ... I'm in the resolutions department. Carry on :) |
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| 20:33.50 | denysonique_ | [TK]D-Fender, right, so the only way to achieve this is via AMI? |
| 20:42.28 | [TK]D-Fender | denysonique_, Did I say "no"? |
| 20:42.37 | [TK]D-Fender | I asked you want the INSTRUCTIONS for it said |
| 20:43.02 | [TK]D-Fender | And you left without answering as to having looked or confirmed anything |
| 20:43.30 | [TK]D-Fender | As for AMI, what does this solve? How you are implying it's use as being a solution? |
| 20:43.42 | [TK]D-Fender | What approach? |
| 20:44.48 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration |
| 20:45.06 | [TK]D-Fender | I linked it before, I'll link it again... |
| 20:45.13 | [TK]D-Fender | Give it a proper read and tell me what you see |
| 20:54.37 | denysonique_ | [TK]D-Fender, I see that in confbridge.conf I can specify conference profiles, user profiles and conference menus. The conference menu options are not sufficient to achieve the goal |
| 20:55.11 | [TK]D-Fender | Conference Menu Configuration Options <--- look AGAIN |
| 20:55.31 | [TK]D-Fender | dialplan_exec |
| 20:55.31 | [TK]D-Fender | |
| 20:55.31 | [TK]D-Fender | (context,exten,priority) |
| 20:55.31 | [TK]D-Fender | |
| 20:55.31 | [TK]D-Fender | Allows one to escape from the conference and execute commands in the dialplan. Once the dialplan exits, the user will be put back into the conference. |
| 20:55.33 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 20:55.39 | [TK]D-Fender | execute DIALPLAN |
| 20:55.43 | [TK]D-Fender | Do whatever the hell you want |
| 20:55.58 | [TK]D-Fender | Trigger off another callout to bring that other party in |
| 20:58.59 | denysonique_ | actually there is even an example: *3=dialplan_exec(addcallers,1,1) |
| 20:59.04 | denysonique_ | [TK]D-Fender, thank you very much |
| 20:59.41 | denysonique_ | I am just a littlebit overwhelmed and somehow missed this. My first day setting up asterisk. |
| 21:00.29 | [TK]D-Fender | You skipped the big print labelled "Menu". Yup, pretty substantial fail. BUt we're moving forward now... |
| 21:00.37 | [TK]D-Fender | The answer is ALWAYS more caffiene |
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| 22:58.05 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 22:58.12 | UncleKiwi | hi |
| 22:59.26 | UncleKiwi | i want to know about a solution FXS that I can put a mobile phone sim card or 2 into |
| 22:59.55 | UncleKiwi | can anyone put me on the right track please |
| 23:02.30 | Samot | I am unaware of any FXS gateway with a mobile sim slot. |
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| 23:02.55 | UncleKiwi | im just googling and the exist |
| 23:03.12 | UncleKiwi | i just want to find one that will work nice with asterisk |
| 23:03.45 | UncleKiwi | https://www.aliexpress.com/item/AXE2G4AN-1G2O-PCI-E-Asterisk-card-with-1-GSM-2-FXO-module-GSM-analog-hybrid-card/1738536922.html?spm=2114.40010308.4.25.VIjlNI |
| 23:03.50 | UncleKiwi | here is one |
| 23:04.36 | Samot | I wouldn't know. |
| 23:04.54 | Samot | I don't use 3g/4g sims. |
| 23:06.26 | [TK]D-Fender | On that card the FXO/FXS has nothing to do with the GSM |
| 23:06.42 | [TK]D-Fender | And expecting to find hybrid cards lke that is a waste |
| 23:08.28 | Samot | Guess he didn't like that answer. |
| 23:08.31 | denysonique_ | Asterisk is trully amazing |
| 23:08.53 | denysonique_ | Just realized that I can add a local channel to a conference, in order to e.g. play a file |
| 23:08.58 | [TK]D-Fender | <PROTECTED> |
| 23:09.30 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 23:09.57 | denysonique_ | well, it is a read error, maybe actually there was a read error on his socket ;p |
| 23:10.20 | denysonique_ | something I found on the web: >I get a call from this number several times a day. When you answer it they hang up and it clicks. When I call the number back it says weasels have eaten our phone system and hangs up. |
| 23:10.27 | Samot | I was being sarcastic.. |
| 23:10.37 | denysonique_ | http://whocallsme.com/Phone-Number.aspx/7015579308 |
| 23:10.39 | denysonique_ | etc |
| 23:10.57 | igcewieling | denysonique_: This is not unusual for phones which accept calls from anywhere. |
| 23:11.52 | igcewieling | On Polycom phones you can use voIpProt.SIP.strictUserValidation="1" to disable that. |
| 23:13.34 | denysonique_ | hmm, does anyone know any fun public PBXes to dail into with some intereseting/funny IVRs? |
| 23:13.39 | UncleKiwi | Samot i was meaning FXO sorry |
| 23:16.32 | denysonique_ | Is there any opensource/free voice tts that works out of the box with asterisk, the quality doesn't have to be to hight, mainly for testing purposes |
| 23:16.50 | [TK]D-Fender | There is never any such thing as "out of the box |
| 23:17.22 | UncleKiwi | ok thanks |
| 23:18.02 | [TK]D-Fender | free = festival. Cheap but better = cepstral |
| 23:19.21 | denysonique_ | or alternatively, how can I execute a command and have it return a sound stream. Is there a solution that isn't forking app_mp3.c? |
| 23:19.51 | denysonique_ | [TK]D-Fender, okay I will check festival out |
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| 23:35.27 | drmessano | GoogleTTS sounds FAR better |
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